EP3544003B1 - Dispositif et procédé de détermination d'une valeur d'évaluation - Google Patents

Dispositif et procédé de détermination d'une valeur d'évaluation Download PDF

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EP3544003B1
EP3544003B1 EP19167397.9A EP19167397A EP3544003B1 EP 3544003 B1 EP3544003 B1 EP 3544003B1 EP 19167397 A EP19167397 A EP 19167397A EP 3544003 B1 EP3544003 B1 EP 3544003B1
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energy
band
measure
signal
distribution
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EP3544003A1 (fr
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Michael Schug
Johannes Hilpert
Stefan Geyersberger
Max Neuendorf
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/022Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
    • G10L19/025Detection of transients or attacks for time/frequency resolution switching
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/002Dynamic bit allocation
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/03Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters

Definitions

  • the present invention relates to coders for coding a signal comprising audio and / or video information, and more particularly to estimating a need for information units for coding this signal.
  • An audio signal to be coded is fed in at an input 1000. This is first fed to a scaling stage 1002, in which what is known as AAC gain control is carried out in order to determine the level of the audio signal. Page information from the scaling is fed to a bitstream formatter 1004, as shown by the arrow between block 1002 and block 1004. The scaled audio signal is then fed to an MDCT filter bank 1006. In the AAC coder, the filter bank implements a modified discrete cosine transform with 50% overlapping windows, the window length being determined by a block 1008.
  • block 1008 is provided for windowing transient signals with shorter windows and windowing more stationary signals with longer windows.
  • the purpose of this is that due to the shorter window for transient signals a higher time resolution is achieved (at the expense of the frequency resolution), while for more stationary signals a higher frequency resolution (at the expense of the time resolution) through longer windows is achieved, with longer windows tending to be preferred because they promise a greater coding gain.
  • each sub-band signal having a certain limited bandwidth that is determined by the corresponding sub-band channel in the filter bank 1006 is determined, and each sub-band signal has a certain number of sub-band samples.
  • the filter bank outputs successive blocks of MDCT spectral coefficients viewed over time, which, generally speaking, represent successive short-term spectra of the audio signal to be encoded at input 1000.
  • TNS temporal noise shaping
  • the TNS technique is used to shape the temporal shape of the quantization noise within each window of the transformation. This is achieved by applying a filtering process to parts of the spectral data of each channel.
  • the coding is done on a window basis.
  • the following steps are carried out in order to use the TNS tool on a window of spectral data, i.e. on a block of spectral values.
  • a frequency range is selected for the TNS tool.
  • a suitable choice is to cover a frequency range from 1.5 kHz up to the highest possible scale factor band with a filter. It should be noted that this frequency range depends on the sampling rate as specified in the AAC standard (ISO / IEC 14496-3: 2001 (E)).
  • LPC linear predictive coding
  • the expected prediction gain PG is obtained as the result of the LPC calculation. Furthermore, the reflection coefficients or Parcor coefficients are obtained.
  • the TNS tool is not used. In this case, control information is written into the bit stream so that a decoder knows that TNS processing has not been carried out.
  • TNS processing is applied.
  • the reflection coefficients are quantized.
  • the order of the noise shaping filter used is determined by removing all reflection coefficients with an absolute value less than a threshold from the "tail" of the reflection coefficient array. The number of remaining reflection coefficients is in the order of magnitude of the noise shaping filter.
  • a suitable threshold is 0.1.
  • the remaining reflection coefficients are typically converted to linear prediction coefficients, a technique also known as the "step-up" procedure.
  • the calculated LPC coefficients are then used as encoder noise shaping filter coefficients, that is to say as prediction filter coefficients.
  • This FIR filter is guided over the specified target frequency range.
  • An autoregressive filter is used for decoding, while a so-called moving average filter is used for coding.
  • the page information for the TNS tool is also fed to the bitstream formatter, as shown by the arrow between the block TNS processing 1010 and the bitstream formatter 1004 in FIG Fig. 3 is shown.
  • Fig. 3 Run through optional tools, not shown, such as a long-term prediction tool, an intensity / coupling tool, a prediction tool, a noise substitution tool, until finally a middle / side encoder 1012 is reached.
  • the middle / side encoder 1012 is active when the audio signal to be encoded is a multi-channel signal, that is to say a stereo signal with a left channel and a right channel. So far, i.e. in the processing direction before block 1012 in Fig. 3 were the left and right stereo channels processed separately from each other, i.e. scaled, transformed through the filter bank, subjected to TNS processing or not, etc.
  • middle / side coder it is then first checked whether middle / side coding makes sense, that is, whether there is any coding gain at all.
  • a middle / side coding will bring a coding gain if the left and right channels are more similar, because then the middle channel, i.e. the sum of the left and right channels, is almost the same as the left or right channel, apart from the scaling by the factor 1/2, while the side channel has only very small values, since it is equal to the difference between the left and right channels.
  • a psychoacoustic model 1020 supplies the quantizer 1014 with a permitted disturbance per scale factor band.
  • the quantizer works iteratively, ie an outer iteration loop is first called, which then calls an inner iteration loop.
  • an outer iteration loop is first called, which then calls an inner iteration loop.
  • a block of values is first quantized at the input of quantizer 1014.
  • the inner loop quantizes the MDCT coefficients, consuming a certain number of bits.
  • the outer loop calculates the distortion and modified energy of the coefficients using the scale factor to call an inner loop again. This process is iterated until a certain set of conditions is reached is satisfied.
  • the signal is reconstructed in order to calculate the disturbance introduced by the quantization and to compare it with the permitted disturbance supplied by the psycho-acoustic model 1020. Furthermore, the scale factors are increased by one level from iteration to iteration, specifically for each iteration of the outer iteration loop.
  • the iteration becomes the analysis-through-synthesis method is terminated, and the scale factors obtained are encoded, as is carried out in block 1014, and supplied in encoded form to the bitstream formatter 1004, as indicated by the arrow between block 1014 and the Block 1004 is drawn.
  • the quantized values are then fed to the entropy coder 1016, which typically entropy encodes using several Huffman code tables for different scale factor bands to translate the quantized values into a binary format.
  • entropy coding in the form of Huffman coding, code tables are used which are created on the basis of expected signal statistics, and in which frequently occurring values are given shorter code words than less frequently occurring values.
  • the entropy-coded values are then also fed to the bit stream formatter 1004 as the actual main information, which then outputs the coded audio signal on the output side according to a specific bit stream syntax.
  • the data reduction of audio signals has become a well-known technique that is the subject of a number of international standards (e.g. ISO / MPEG-1, MPEG-2 AAC, MPEG-4).
  • the input signal is brought into a compact, data-reduced representation by means of a so-called encoder using perception-related effects (psychoacoustics, psychooptics).
  • a spectral analysis of the signal is usually carried out and the corresponding signal components are quantized, taking into account a perception model, and then coded as a so-called bit stream in the most compact way possible.
  • So-called perceptual entropy can be used to estimate before the actual quantization how many bits a certain section of the signal to be coded will need.
  • the PE also provides a measure of how difficult it is for the encoder to encode a particular signal or parts of it.
  • the perceptual entropy or any estimated value for a need for information units for coding a signal can be used to estimate whether the signal is transient or stationary, since transient signals also require more bits for coding than stationary signals.
  • the estimation of a transient property of a signal is used, for example, to make a window length decision as indicated by block 1008 in Fig. 3 is indicated to perform.
  • Fig. 6 the perceptual entropy calculated according to ISO / IEC IS 13818-7 (MPEG-2 advanced audio coding (AAC)) is shown.
  • AAC MPEG-2 advanced audio coding
  • the in Fig. 6 is used.
  • the parameter pe stands for the perceptual entropy.
  • width (b) stands for the number of spectral coefficients in the respective band b.
  • e (b) is the energy of the signal in this band.
  • nb (b) is the matching masking threshold or, in more general terms, the permitted interference that can be introduced into the signal, for example by quantization, so that a human listener still hears no or only a negligible interference.
  • the bands can be derived from the band division of the psychoacoustic model (block 1020 in Fig. 3 ) or the so-called scale factor bands (scfb) used in the quantization.
  • the psychoacoustic masking threshold is the energy value that the quantization error should not exceed.
  • FIG. 6 The figure shown thus shows how well a perceptual entropy determined in this way works as an estimate for the number of bits required for coding.
  • the respective perceptual entropy was plotted for each individual block as a function of the bits used.
  • the test piece used contains a typical mixture of music, language and individual instruments.
  • the points would congregate along a straight line through the zero point.
  • the expansion of the point sequence with the deviations from the ideal line illustrates the imprecise estimate.
  • the disadvantage of the in Fig. 6 The concept shown is the deviation that expresses itself to the effect that, for example, the value for the perceptual entropy is too large, which in turn means that the quantizer is signaled that more bits are required than actually required. This has the result that the quantizer quantizes too finely, that is to say that it does not exhaust the degree of permitted interference, which results in a reduced coding gain.
  • the quantizer is signaled that fewer bits than actually required are required to encode the signal. This in turn has the consequence that the quantizer quantizes too roughly, which would immediately lead to an audible disturbance in the signal, unless countermeasures are taken.
  • the countermeasures can be that the quantizer still needs one or more further iteration loops, which increases the computing time of the encoder.
  • FIG. 8 Another calculation of the perceptual entropy, which is very time consuming, is in Fig. 8 shown.
  • Fig. 8 the case is shown in which the perceptual entropy is calculated line by line.
  • the disadvantage is the higher computational effort involved in the line-by-line calculation.
  • spectral coefficients X (k) are used, where kOffset (b) denotes the first index of band b.
  • the US 2002/103637 A1 discloses a concept for improving the performance of coding systems employing radio frequency reconstruction techniques. For this purpose, a coding difficulty or a measure for the workload of an encoder is calculated on the encoder side in order to control the crossover frequency as a function of this, which determines the frequency up to which a signal is encoded with a source encoder, the proportion of the signal is encoded above the crossover frequency by a high frequency reconstruction method.
  • the Perceptual Entropy is calculated, which is based on a spectral value being squared and then weighted with a number equal to the number of lines in the current band divided by the psychoacoustic threshold for this Band, and then take the logarithm of the result. Summing up all such logarithms in a band then gives the perceptual entropy in this band.
  • a distortion energy can also be calculated at the end of the source coding process by adding up the distortion energy in each band and weighting it with a loudness curve.
  • the object of the present invention is to create an efficient and yet precise concept for determining an estimated value for a requirement of information units for coding a signal.
  • the present invention is based on the knowledge that a frequency band-wise calculation of the estimated value for a requirement for information units for reasons of computing time It should be noted, however, that in order to obtain an accurate estimate of the estimate, the distribution the energy in the frequency band that is to be calculated band by band must be taken into account.
  • the entropy coder following the quantizer is thus implicitly “drawn into” the determination of the estimated value for the requirement of information units.
  • the entropy coding enables a smaller number of bits to be required for the transmission of smaller spectral values than for the transmission of larger spectral values.
  • the entropy coder is particularly efficient when spectral values quantized to zero can be transmitted. Since these will typically occur most frequently, the code word for transmitting a spectral line quantized to zero is the shortest code word, and the code word for transmitting an ever larger quantized spectral line is always longer.
  • the measure for the distribution of the energy in the frequency band can be determined on the basis of the actual amplitudes, or by estimating the frequency lines that are not quantized to zero by the quantizer.
  • This dimension which is also referred to as “nl”, where nl stands for “number of active lines”, that is to say for the number of active lines, is preferred for reasons of computing time efficiency.
  • the number of spectral lines quantized to zero or a finer subdivision can also be taken into account, this estimation becoming more and more precise the more information from the downstream entropy coder is taken into account.
  • the entropy coder is built on the basis of Huffman code tables, properties of these code tables can be integrated particularly well, since the code tables are not calculated on-line based on the signal statistics, but rather because the code tables are fixed anyway, regardless of the actual signal.
  • the measure for the distribution of the energy in the frequency band is carried out by determining the lines that still survive after the quantization, that is to say the number of active lines.
  • the present invention is advantageous in that an estimated value for a need for information content is determined which is on the one hand more precise and on the other hand more efficient than in the prior art.
  • the present invention can be scaled for various applications, since, depending on the desired accuracy of the estimated value, more and more properties of the entropy coder can be included in the estimation of the bit requirement, but at the cost of increased computing time.
  • the signal which can be an audio and / or a video signal, is fed in via an input 100.
  • the signal is preferably already available as a spectral representation with spectral values. However, this is not absolutely necessary, since some calculations can also be carried out with a time signal by means of appropriate bandpass filtering, for example.
  • the signal is fed to a device 102 for providing a measure of an allowable interference for a frequency band of the signal.
  • the permitted disturbance can, for example, by means of a psycho-acoustic model, as it is based on Fig. 3 (Block 1020) has to be determined.
  • the means 102 is also effective to provide a measure of the energy of the signal in the frequency band.
  • a prerequisite for a band-by-band calculation is that a frequency band for which a permitted interference or signal energy is specified contains at least two or more spectral lines of the spectral representation of the signal.
  • the frequency band will preferably be a scale factor band because the bit requirement estimation is required directly by the quantizer in order to determine whether a quantization that has taken place fulfills a bit criterion or not.
  • the device 102 is designed to feed both the permitted interference nb (b) and the signal energy e (b) of the signal in the band to a device 104 for calculating the estimated value for the requirement for bits.
  • the means 104 for calculating the estimated value for the requirement of bits is designed to take into account, in addition to the permitted interference and the signal energy, a measure nl (b) for a distribution of the energy in the frequency band, the distribution of the energy in the frequency band from deviates from a completely even distribution.
  • the measure for the distribution of the energy is calculated in a device 106, the device 106 requiring at least one band, namely the considered frequency band of the audio or video signal either as a bandpass signal or directly as a sequence of spectral lines, e.g. to be able to perform a spectral analysis of the band in order to obtain the measure for the distribution of the energies in the frequency band.
  • the audio or video signal can be fed to the device 106 as a time signal, the device 106 then performing band filtering and an analysis in the band.
  • the audio or video signal that is fed to device 106 can already be present in the frequency range, such as, for example, as MDCT coefficients, or as a bandpass signal in the filter bank with a smaller number of bandpasses compared to an MDCT filter bank -Filter.
  • the means 106 for calculating is designed to take into account current amounts of spectral values in the frequency band for calculating the estimated value.
  • the device for calculating the measure for the distribution of the energy can be designed to determine a number of spectral values as a measure for the distribution of the energy, the amount of which is greater than or equal to a predetermined amount threshold, or the amount of which is less than or equal to the amount threshold , the absolute value threshold preferably being an estimated quantizer stage which, in a quantizer, causes values less than or equal to the quantizer stage to be quantized to zero.
  • the measure for the energy is the number of active lines, i.e. the number of lines that survive or are not equal to zero after quantization.
  • Fig. 2a shows a preferred embodiment for the means 106 for calculating the measure for the distribution of the energy in the frequency band.
  • the measure of the distribution of energy in the frequency band is in Fig. 2a denoted by nl (b).
  • the form factor ffac (b) is already a measure of the distribution of energy in the frequency band.
  • the measure for the spectral distribution nl is obtained from the form factor ffac (b) by weighting with the 4th root of the signal energy e (b) divided by the bandwidth width (b) or number of lines determined in the scale factor band b.
  • nl (b) is an example of is a quantity that represents an estimate of the number of lines relevant for quantization.
  • the form factor ffac (b) is calculated by forming the absolute value of a spectral line and then taking the root of this spectral line and then adding up the "rooted" amounts of the spectral lines in the band.
  • Figure 2b shows a preferred embodiment of the means 104 for calculating the estimated value pe, where in Figure 2b Another case distinction is introduced, namely when the logarithm to base 2 of the ratio of the energy to the permitted disturbance is greater than a constant factor c1 or equal to the constant factor.
  • the above alternative in block 104 is used, that is to say the measure for the spectral distribution nl is multiplied by the logarithm expression.
  • Figure 4a a band in which there are four spectral lines, all of the same size. The energy in this band is thus evenly distributed over the band.
  • Figure 4b a situation where the energy in the band resides in one spectral line while the other three spectral lines are the same are zero.
  • the band shown could be before quantization or could be obtained after quantization if the in Figure 4b Spectral lines set to zero before the quantization are smaller than the first quantizer stage and are thus set to zero by the quantizer, ie they do not "survive".
  • the number of active lines in Figure 4b is therefore equal to 1, with the parameter nl in Figure 4b to the square root of 2 is calculated.
  • the value nl i.e. the measure for the spectral distribution of the energy in Figure 4a calculated to 4. This means that the spectral distribution of the energy is more uniform when the measure for the distribution of the spectral energy is larger.
  • the in Figure 4b The case shown can be coded with only one relevant line with fewer bits, since the three spectral lines set to zero can be transmitted very efficiently.
  • the simpler quantizability of the in Figure 4b The case shown is based on the fact that after the quantization and lossless coding, smaller values and, in particular, values quantized to zero require fewer bits for transmission.
  • Fig. 2a The form factor shown is also required elsewhere in the encoder, for example within the quantization block 1014 for determining the quantization step size. If the form factor is already calculated elsewhere, it does not have to be recalculated for bit estimation, so that the inventive concept for improved estimation of the measure for the required bits manages with a minimum of additional computing effort.
  • X (k) is the spectral coefficient to be quantized later, while the variable kOffset (b) designates the first index in band b.
  • the new formula for calculating an improved band-wise perceptual entropy is based on the multiplication of the measure for the spectral distribution of the energy and the logarithm expression in which the signal energy e (b) occurs in the numerator and the permitted disturbance in the denominator, depending on requirements a term inserted within the logarithm can be, as it is already in Fig. 7 is shown. This term can, for example, also be 1.5, but can also be equal to zero, as in the in Figure 2b case shown, this being e.g. B. can be determined empirically.
  • Fig. 5 indicated, from which the perceptual entropy calculated according to the invention can be seen, plotted over the required bits. A higher accuracy of the estimation compared to the comparative examples in the Fig. 6 , 7th and 8th can be clearly seen.
  • the modified band-by-band calculation according to the invention also performs at least equally as compared to the line-by-line calculation.
  • the method according to the invention can be implemented in hardware or in software.
  • the implementation can take place on a digital storage medium, in particular a floppy disk or CD with electronically readable control signals which can interact with a programmable computer system in such a way that the method is carried out.
  • the invention thus also consists in a computer program product with a program code stored on a machine-readable carrier for carrying out the method according to the invention when the computer program product runs on a computer.
  • the invention can thus be implemented as a computer program with a program code for carrying out the method when the computer program runs on a computer.

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Claims (13)

  1. Dispositif pour déterminer une valeur estimée d'un besoin d'unités d'information pour coder un signal qui présente des informations audio ou vidéo, dans lequel le signal présente plusieurs bandes de fréquences, aux caractéristiques suivantes:
    un dispositif (102) destiné à fournir une mesure pour une interférence autorisée pour une bande de fréquences du signal, où la bande de fréquences comporte au moins deux valeurs spectrales d'une représentation spectrale du signal, et une mesure pour une énergie du signal dans la bande de fréquences;
    caractérisé par
    un moyen (106) destiné à calculer une mesure pour une répartition de l'énergie dans la bande de fréquences, où la répartition de l'énergie dans la bande de fréquences s'écarte d'une répartition complètement uniforme; et
    un moyen (104) destiné à calculer la valeur estimée à l'aide de la mesure pour l'interférence, de la mesure pour l'énergie et de la mesure pour la répartition de l'énergie.
  2. Dispositif selon la revendication 1, dans lequel le moyen (106) destiné à calculer est conçu pour tenir compte, pour calculer la mesure pour la répartition de l'énergie, des quantités de valeurs spectrales dans la bande de fréquences.
  3. Dispositif selon la revendication 1 ou 2, dans lequel le moyen (106) destiné à calculer la mesure pour la répartition de l'énergie est conçu pour déterminer, comme mesure pour la répartition de l'énergie, un nombre de valeurs spectrales dont la quantité est supérieure ou égale à un seuil de quantité prédéterminé, ou dont la quantité est inférieure ou égale au seuil de quantité.
  4. Dispositif selon la revendication 3, dans lequel le seuil de quantité est un étage de quantificateur exact ou estimé qui, dans un quantificateur, fait que des valeurs inférieures ou égales à l'étage de quantificateur soient quantifiées à zéro.
  5. Dispositif selon l'une des revendications précédentes, dans lequel le moyen (106) destiné à calculer est conçu pour calculer un facteur de forme selon l'équation suivante: ffac b = k = kOffset b kOffset b + 1 1 X k ,
    Figure imgb0015
    où X(k) est une valeur spectrale à un indice de fréquence k, où kOffset est une première valeur spectrale dans une bande b, et où ffac(b) est le facteur de forme.
  6. Dispositif selon l'une des revendications précédentes,
    dans lequel le moyen (106) destiné à calculer est conçu pour tenir compte d'une quatrième racine d'un rapport entre l'énergie dans la bande de fréquences et une largeur de la bande de fréquences ou le nombre de valeurs spectrales dans la bande de fréquences.
  7. Dispositif selon l'une des revendications précédentes,
    dans lequel le moyen (106) destiné à calculer est conçu pour calculer la mesure pour la répartition de l'énergie selon les équations suivantes: nl b = ffac b e b width b 0.25
    Figure imgb0016
    ffac b = k = kOffset b kOffset b + 1 1 X k ,
    Figure imgb0017
    où X (k) est une valeur spectrale à un indice de fréquence k, où kOffset est une première valeur spectrale dans une bande b, où ffac(b) est un facteur de forme, où nl(b) représente la mesure pour la répartition d'énergie dans la bande b, où e(b) est une énergie de signal dans la bande b, et où width(b) est une largeur de la bande.
  8. Dispositif selon l'une des revendications précédentes,
    dans lequel le moyen (104) destiné à calculer la valeur estimée est conçu pour utiliser un quotient de l'énergie dans la bande de fréquences et de l'interférence dans la bande de fréquences.
  9. Dispositif selon l'une des revendications précédentes,
    dans lequel le moyen (104) destiné à calculer la valeur estimée est conçu pour calculer la valeur estimée à l'aide de l'expression suivante: pe = b nl b log 2 e b nb b + s
    Figure imgb0018
    où pe est la valeur estimée, où nl(b) représente la mesure pour la répartition de l'énergie dans la bande b, où e(b) est une énergie du signal dans la bande b, où nb(b) est l'interférence autorisée dans la bande b est, et où s est un terme additif, qui est de préférence égal à 1,5.
  10. Dispositif selon l'une des revendications précédentes,
    dans lequel le moyen (104) destiné à calculer la valeur estimée est conçu pour calculer la valeur estimée selon l'équation suivante: pe = b nl b log 2 e b nb b + s
    Figure imgb0019
    où est d'application: nl b = ffac b e b width b 0.25 ,
    Figure imgb0020
    et
    où est d'application: ffac b = k = kOffset b kOffset b + 1 1 X k ,
    Figure imgb0021
    où pe est la valeur estimée, où nl(b) représente la mesure pour la répartition de l'énergie dans la bande b, où e(b) est une énergie du signal dans la bande b, où nb(b) est l'interférence autorisée dans la bande b, où s est un terme additif, qui est de préférence égal à 1,5, où X (k) est une valeur spectrale à un indice de fréquence k, où kOffset est une première valeur spectrale dans une bande b, où ffac(b) est un facteur de forme, et où width(b) est une largeur de la bande.
  11. Dispositif selon l'une des revendications précédentes,
    dans lequel le signal est donné sous forme de représentation spectrale avec des valeurs spectrales.
  12. Procédé de détermination d'une valeur estimée d'un besoin d'unités d'informations pour coder un signal qui présente des informations audio ou vidéo, dans lequel le signal présente plusieurs bandes de fréquences, aux étapes suivantes consistant à:
    fournir (102) une mesure pour une interférence autorisée pour une bande de fréquences du signal, où la bande de fréquences comporte au moins deux valeurs spectrales d'une représentation spectrale du signal, et une mesure pour une énergie du signal dans la bande de fréquences;
    caractérisé par le fait de
    calculer (106) une mesure pour une répartition de l'énergie dans la bande de fréquences, où la répartition de l'énergie dans la bande de fréquences s'écarte d'une répartition complètement uniforme; et
    calculer (104) la valeur estimée à l'aide de la mesure pour l'interférence, de la mesure pour l'énergie et de la mesure pour la répartition de l'énergie.
  13. Programme d'ordinateur avec un code de programme pour réaliser le procédé de détermination d'une valeur estimée pour un besoin d'unités d'information pour coder un signal selon la revendication 12 lorsque le programme est exécuté sur un ordinateur.
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US7318028B2 (en) 2008-01-08
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