EP3020043B1 - Facteur d'échelle optimisé pour l'extension de bande de fréquence dans un décodeur de signaux audiofréquences - Google Patents

Facteur d'échelle optimisé pour l'extension de bande de fréquence dans un décodeur de signaux audiofréquences Download PDF

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EP3020043B1
EP3020043B1 EP14749907.3A EP14749907A EP3020043B1 EP 3020043 B1 EP3020043 B1 EP 3020043B1 EP 14749907 A EP14749907 A EP 14749907A EP 3020043 B1 EP3020043 B1 EP 3020043B1
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frequency
filter
band
frequency band
signal
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EP3020043A1 (fr
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Magdalena KANIEWSKA
Stéphane RAGOT
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Koninklijke Philips NV
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Orange SA
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech 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
    • G10L19/04—Speech 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 predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/087—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters using mixed excitation models, e.g. MELP, MBE, split band LPC or HVXC
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/48—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use
    • G10L25/72—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use for transmitting results of analysis
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/038—Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech 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
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech 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
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech 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
    • G10L19/02—Speech 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech 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
    • G10L19/04—Speech 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 predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding

Definitions

  • the present invention relates to the field of coding / decoding and audio-frequency signal processing (such as speech, music or other signals) for their transmission or storage.
  • the invention relates to a method and an apparatus for determining an optimized scale factor for adjusting the level of an excitation signal or, in a similar manner, a filter during a band extension. frequency in a decoder or a processor performing audio-frequency signal enhancement.
  • state of the art audio signal coding (mono) consists of perceptual encoding by transform or subband, with parametric coding of high frequencies by tape replication.
  • a review of conventional speech and audio coding methods can be found in the books WB Kleijn and KK Paliwal (Eds.), Speech Coding and Synthesis, Elsevier, 1995 ; M. Bosi, RE Goldberg, Introduction to Digital Audio Coding and Standards, Springer 2002 ; J. Benesty, MM Sondhi, Y. Huang (Eds.), Handbook of Speech Processing, Springer 2008 .
  • 3GPP AMR-WB Adaptive Multi-Rate Wideband codec (decoder and decoder), which operates at an input / output frequency of 16 kHz and in which the signal is divided into two sub-bands, the low band (0-6.4 kHz) which is sampled at 12.8 kHz and coded by CELP model and the high band (6.4-7 kHz) which is parametrically reconstructed by " band extension " ( or BWE for "Bandwidth Extension” with or without additional information depending on the mode of the current frame.
  • band extension or BWE for "Bandwidth Extension
  • the limitation of the coded band of the AMR-WB codec at 7 kHz is essentially related to the fact that the transmit frequency response of the broadband terminals has been approximated at the time of standardization (ETSI / 3GPP then ITU-T T) according to the frequency mask defined in the ITU-T P.341 standard and more precisely by using a so-called "P341" filter defined in the ITU-T G.191 standard. which cuts frequencies above 7 kHz (this filter respects the mask defined in P.341).
  • a signal sampled at 16 kHz may have a defined audio band of 0 to 8000 Hz; the AMR-WB codec thus introduces a limitation of the high band in comparison with the theoretical bandwidth of 8 kHz.
  • the 3GPP AMR-WB speech codec was standardized in 2001 mainly for circuit-mode (CS) telephony applications over GSM (2G) and UMTS (3G). This same codec was also standardized in 2003 in the ITU-T as Recommendation G.722.2 "Wideband coding speech at around 16kbit / s using Adaptive Multi-Rate Wideband (AMR-WB)".
  • the principle of band extension in the AMR-WB codec is rather rudimentary. Indeed, the high band (6.4-7 kHz) is generated by formatting a white noise through a temporal envelope (applied in the form of gains per subframe) and frequency (by the application of a linear prediction synthesis filter or LPC for "Linear Predictive Coding").
  • This band extension technique is illustrated in figure 1 .
  • the present invention improves the situation.
  • an additional filter of a lower order than the filter of the first frequency band to be equalized makes it possible to avoid overestimations of energy in the high frequencies which could result from local fluctuations of the envelope and which can disrupt the equalization of the prediction filters.
  • the band extension method comprises a step of applying the optimized scaling factor to the extended excitation signal.
  • the application of the optimized scaling factor is combined with the filtering step in the second frequency band.
  • the coefficients of the additional filter are obtained by truncation of the transfer function of the linear prediction filter of the first frequency band to obtain a lower order.
  • the coefficients of the additional filter are modified according to a criterion of stability of the additional filter.
  • the optimized scaling factor is calculated to avoid annoying artifacts that might arise in the event that the higher order filter frequency response of the first band near the common frequency would reveal a peak or a valley. of the signal.
  • additional information can be used to improve the quality of the extended signal for a predetermined mode of operation.
  • the invention relates to a decoder comprising a device as described.
  • It relates to a computer program comprising code instructions for implementing the steps of the method of determining an optimized scale factor as described, when these instructions are executed by a processor.
  • the invention relates to a storage medium, readable by a processor, integrated or not to the device for determining an optimized scaling factor, possibly removable, storing a computer program implementing a method of determining a optimized scaling factor as previously described.
  • the figure 3 illustrates an exemplary decoder, compatible with the AMR-WB / G.722.2 standard, in which there is a band extension comprising a determination of an optimized scale factor according to one embodiment of the method of the invention, implemented implemented by the tape extension device illustrated by block 309.
  • AMR-WB decoding which operates with an output sampling frequency of 16 kHz
  • the CELP decoding (BF for low frequencies) still operates at the internal frequency of 12.8 kHz, as in AMR-WB, and the band extension (HF for high frequencies) used for the invention operates at the frequency of 16 kHz
  • the syntheses BF and HF are combined (block 312) at the frequency fs after adequate resampling (block 306 and internal processing block 311).
  • the combination of the low and high bands may be done at 16 kHz, after resampling the low band of 12.8 to 16 kHz, before resampling the combined signal at the frequency fs .
  • the decoder makes it possible to extend the decoded low band (50-6400 Hz taking into account the high-pass filtering at 50 Hz at the decoder, 0-6400 Hz in the general case) with an extended band whose width varies, ranging from approximately 50-6900 Hz to 50-7700 Hz depending on the mode implemented in the current frame.
  • the extension of the excitation is carried out in the frequency domain in a band of 5000 to 8000 Hz, to allow bandpass filtering of width 6000 to 6900 or 7700 Hz.
  • the HF gain correction information (0.8 kbit / s) transmitted at 23.85 kbit / s is decoded here. Its use is detailed below, with reference to the figure 4 .
  • the high band synthesis part is performed in block 309 representing the band extension device used for the invention and which is detailed in FIG. figure 7 in one embodiment.
  • a delay (block 310) is introduced to synchronize the outputs of the blocks 306 and 307 and the high band synthesized at 16 kHz is resampled from 16 kHz to the fs frequency (block output 311).
  • the value of the delay T depends on how to synthesize the high band signal, the frequency fs as well as the post-processing of the low frequencies. Thus, in general, the value of T in the block 310 will have to be adjusted according to the specific implementation.
  • the low and high bands are then combined (added) in block 312 and the resulting synthesis is post-processed by high-order 50 Hz (type IIR) high-pass filtering whose coefficients depend on the frequency fs (block 313) and output postprocessing with optional "noisegate” application similar to G.718 (block 314).
  • high-order 50 Hz type IIR
  • the block 400 from a decoded excitation signal in a first frequency band u ( n ), performs a band extension to obtain an extended excitation signal u HB ( n ) on at least a second frequency band.
  • the optimized scale factor estimation according to the invention is independent of how to obtain the signal u HB ( n ) .
  • a condition regarding its energy is important, however. Indeed, it is necessary that the energy of the high band of 6000 to 8000 Hz is at a level similar to the energy of the band 4000 to 6000 Hz of the decoded excitation signal at the output of the block 302. since the low-band signal is de-emphasized (block 305), it is also necessary to apply the de-emphasis to the high-band excitation signal, either by using an own de-emphasis filter or by multiplying by a constant factor which corresponds to a mean attenuation of mentioned filter. This condition does not apply to the 23.85 kbit / s rate that uses the additional information transmitted by the encoder. In this case, the energy of the high band excitation signal must be consistent with the signal energy corresponding to the encoder, as explained later.
  • the frequency band extension may for example be implemented in the same way as for the AMR-WB decoder described with reference to FIG. figure 1 in blocks 100 to 102, from a white noise.
  • this band extension can be performed from a combination of a white noise and a decoded excitation signal as illustrated and subsequently described for blocks 700 to 707 of FIG. figure 7 .
  • the tape expansion module may also be independent of the decoder and may extend a band of an existing audio signal stored or transmitted to the expansion module, with an analysis of the audio signal to extract one excitation and a LPC filter.
  • the excitation signal at the input of the extension module is no longer a decoded signal but a signal extracted after analysis, as well as the coefficients of the linear prediction filter of the first frequency band used in the method of determining the optimized scale factor in an implementation of the invention.
  • the determination of the optimized scale factor is also performed by the determination (in 401a) of a linear prediction filter called additional filter, of a lower order than the linear prediction filter of the first frequency band 1 / ⁇ (z ), the coefficients of the additional filter being obtained from the parameters decoded or extracted from the first frequency band.
  • the optimized scaling factor is then calculated (at 401b) based on at least one of these coefficients to be applied to the extended excitation signal u HB ( n ).
  • an extended excitation signal u HB (n) is obtained during a frequency band extension method E601 which comprises a decoding or extraction step, in a first so-called low band frequency band, an excitation signal and parameters of the first frequency band, for example the coefficients of the linear prediction filter of the first frequency band.
  • a step E602 determines a linear prediction filter called additional filter, of a lower order than that of the first frequency band. To determine this filter, the parameters of the first decoded or extracted frequency band are used.
  • this step is performed by truncation of the transfer function of the linear prediction filter of the low band to obtain a lower filter order, for example 2. These coefficients can then be modified according to a criterion of stability as explained above with reference to the figure 4 .
  • a step E603 is implemented to calculate the optimized scale factor to be applied to the extended excitation signal.
  • This optimized scale factor is for example calculated from the frequency response of the additional filter at a common frequency between the low band (first frequency band) and the high band (second frequency band). A minimum value that can be chosen between the frequency response of this filter and those of the low band and high band filters. This avoids the overestimation of energy that could exist in state-of-the-art methods.
  • This step of calculating the optimized scale factor is for example described above with reference to the figure 4 and to Figures 5a and 5b .
  • Step E604 performed by block 402 or 409 (depending on the decoding rate) for the band extension, applies the optimized scaling factor thus calculated to the extended excitation signal so as to obtain an extended excitation signal.
  • the optimized scaling factor device 708 is integrated in a tape expansion device described now with reference to the figure 7 .
  • This optimized scale factor determination device illustrated by block 708 implements the method of determining the optimized scale factor described above with reference to FIG. figure 6 .
  • the band extension block 400 of the figure 4 includes blocks 700 to 707 of the figure 7 described now.
  • a decoded or analytically estimated low band excitation signal is received ( u ( n )).
  • the band extension here uses the decoded excitation at 12.8 kHz (exc2 or u ( n )) at the output of the block 302 of the figure 3 .
  • the generation of the oversampled and extended excitation is carried out in a frequency band ranging from 5 to 8 kHz including a second frequency band (6.4-8kHz) greater than the first band of frequency (0-6.4 kHz).
  • the generation of an extended excitation signal is effected at least on the second frequency band but also on a part of the first frequency band.
  • this signal is transformed to obtain an excitation signal spectrum U ( k ) by the time-frequency transformation module 500.
  • the DCT-IV transformation is implemented by FFT according to the algorithm called " Evolved DCT (EDCT)" described in the article by DM Zhang, HT Li, A Low Complexity Transform - Evolved DCT, IEEE 14th International Conference on Computational Science and Engineering (CSE), Aug. 2011, pp. 144-149 , and implemented in ITU-T G.718 Annex B and G.729.1 Annex E.
  • EDCT Evolved DCT
  • the DCT-IV transformation may be replaced with other short-term time-frequency transformations of the same length and in the field of excitation, such as an FFT (for " Fast Fourier Transform "in English ) or DCT-II ( Discrete Cosine Transform - Type II).
  • FFT Fast Fourier Transform
  • DCT-II Discrete Cosine Transform - Type II
  • MDCT for "Modified Discrete Cosine Tranform" in English.
  • the delay T in the block 310 of the figure 3 should be adjusted (reduced) adequately according to the additional delay due to the analysis / synthesis by this transform.
  • This approach preserves the original spectrum in this band and avoids introducing distortions in the 5000-6000 Hz band during the addition of HF synthesis with BF synthesis - particularly the signal phase (implicitly represented in the DCT-IV domain) in this band is preserved.
  • the band 6000-8000 Hz of U HB 1 ( k ) is here defined by copying the 4000-6000 Hz band of U ( k ) since the value of start_band is preferably fixed at 160.
  • the value of start_band can be made adaptive around the value of 160.
  • the details of the adaptation of the value start_band are not described here because they go beyond the scope of the invention without changing the scope.
  • the noise generation block 702 For some broadband signals (sampled at 16 kHz), the high band (> 6 kHz) may be noisy, harmonic or have a mixture of noise and harmonics. In addition, the level of harmonicity in the 6000-8000 Hz band is generally correlated with that of the lower frequency bands.
  • the noise in the 6000-8000 Hz band
  • U HBN k - 1 + 13849 k 240 , ⁇ , 319 with the convention that U HBN (239) in the current frame corresponds to the value U HBN (319) of the previous frame.
  • this noise generation can be replaced by other methods.
  • the combination block 703 can be realized in different ways.
  • G HBN is a normalization factor for equalizing the energy level between the two signals
  • the coefficient ⁇ (between 0 and 1) is adjusted according to parameters estimated from the decoded low band and the coefficient ⁇ (between 0 and 1) depends on ⁇ .
  • N ( k 1 , k 2 ) is the set of indices k for which the index coefficient k is classified as being associated with noise.
  • This set can be obtained for example by detecting the local peaks in U '( k ) verifying
  • and considering that these lines are not associated with noise, ie (by applying the negation of the previous condition): NOT at b at ⁇ k ⁇ b
  • other methods of calculating the noise energy are possible, for example by taking the median value of the spectrum on the band in question or by applying a smoothing to each frequency line before calculating the energy per band.
  • the calculation of ⁇ may be replaced by other methods.
  • different parameters by limiting its value between 0 and 1.
  • the factors ⁇ and ⁇ may be adapted to take account of the fact that noise injected into a given band of the signal is generally perceived as stronger than a harmonic signal at the same energy in the same band.
  • block 703 realizes the equivalent of block 101 of the figure 1 to normalize the white noise according to an excitation which is on the other hand here in the frequency domain, already extended to the rate of 16 kHz; in addition, the mix is limited to the band 6000-8000 Hz.
  • an embodiment of block 703 may be considered, where the spectra, U HB 1 ( k ) or G HBN U HBN ( k ), are selected (switched) adaptively, which amounts to allowing only the values 0 or 1 for ⁇ ; this approach amounts to classifying the type of excitation to be generated in the 6000-8000 Hz band
  • the block 704 optionally carries out a dual operation of application of bandpass filter frequency response and deemphasis filtering (or deemphasis) in the frequency domain.
  • the deemphasis filtering may be performed in the time domain, after block 705 or even before block 700; however, in this case, bandpass filtering performed in block 704 may leave some low frequency components of very low levels that are amplified by de-emphasis, which may slightly discern the decoded low band. For this reason, it is preferred here to perform the deemphasis in the frequency domain.
  • the HF synthesis is not de-emphasized.
  • the high frequency signal is on the contrary de-emphasized so as to bring it back into a domain consistent with the low signal. frequencies (0-6.4 kHz) coming out of block 305 of the figure 3 . This is important for the estimation and subsequent adjustment of the energy of the HF synthesis.
  • the de-emphasis can be performed in an equivalent way in the time domain after inverse DCT.
  • band-pass filtering is applied with two separate parts: one fixed high-pass, the other adaptive low-pass (flow-rate function).
  • This filtering is performed in the frequency domain.
  • bandpass filtering can be adapted by defining a single filtering step combining the high-pass and low-pass filtering.
  • the bandpass filtering may be performed in an equivalent manner in the time domain (as in block 112 of the present invention). figure 1 ) with different filter coefficients according to the flow rate, after a reverse DCT step.
  • this step it is advantageous to carry out this step directly in the frequency domain because the filtering is carried out in the field of LPC excitation and therefore the problems of circular convolution and edge effects are very limited in this field. .
  • This excitation sampled at 16 kHz is then optionally scaled by gains defined by subframe of 80 samples (block 707).
  • the realization of the block 706 differs from that of the block 101 of the figure 1 because the energy at the current frame is taken into account in addition to that of the sub-frame. This makes it possible to have the ratio of the energy of each sub-frame with respect to the energy of the frame. Energy ratios (or relative energies) are compared rather than the absolute energies between low band and high band.
  • this scaling step makes it possible to keep in the high band the energy ratio between the subframe and the frame in the same way as in the low band.
  • the block 708 then performs a scaling factor calculation by subframe of the signal (steps E602 to E 603 of the figure 6 ), as previously described with reference to the figure 6 and detailed in figure 4 and 5 .
  • this filtering can be done in the same way as that described for block 111 of the figure 1 of the AMR-WB decoder, however the order of the filter goes to 20 at the rate of 6.6, which does not significantly change the quality of the synthesized signal.
  • the step of filtering by a linear prediction filter 710 for the second frequency band is combined with the application of the optimized scaling factor, which reduces the processing complexity.
  • the filtering steps 1 / ⁇ (z / ⁇ ) and the application of the optimized scaling factor g HB 2 are combined with a single filtering step g HB 2 / ⁇ (z / ⁇ ) to reduce the processing complexity. .
  • the coding of the low band (0-6.4 kHz) may be replaced by a CELP coder other than that used in AMR-WB, for example the CELP coder in G.718 to 8. kbit / s.
  • a CELP coder other than that used in AMR-WB, for example the CELP coder in G.718 to 8. kbit / s.
  • other encoders in wide band or operating at frequencies higher than 16 kHz in which the coding of the low band operates at an internal frequency at 12.8 kHz could be used.
  • the invention can be obviously adapted to other sampling frequencies than 12.8 kHz, when a low frequency encoder operates at a sampling frequency lower than that of the original or reconstructed signal.
  • the low band decoding does not use a linear prediction, it does not have an excitation signal to be extended, in this case it will be possible to carry out an LPC analysis of the reconstructed signal in the current frame and calculate an LPC excitation. so as to be able to apply the invention.
  • the excitation ( u ( n )) is resampled, for example by linear interpolation or "spline" cubic, from 12.8 to 16 kHz before transformation (for example DCT-IV)
  • This variant has the defect of being more complex, because the transform (DCT-IV) of the excitation is then calculated over a greater length and the resampling is not carried out in the field of the transformed.
  • all the calculations necessary for the estimation of the gains can be carried out in a logarithmic domain.
  • the low band excitation u ( n ) and the LPC 1 / ⁇ (z) filter will be estimated per frame, by LPC analysis of a low band signal whose band must be extended.
  • the low band excitation signal is then extracted by analyzing the audio signal.
  • the low band audio signal is resampled before the excitation extraction step, so that the excitation extracted from the audio signal (by linear prediction) is already resolved. sampled.
  • the illustrated tape extension at figure 7 applies in this case to a low band which is not decoded but analyzed.
  • the figure 8 represents an embodiment of a device for determining an optimized scale factor 800 according to the invention. This may be an integral part of an audio-frequency signal decoder or equipment receiving decoded or non-decoded audio signals.
  • This type of device comprises a PROC processor cooperating with a memory block BM having a memory storage and / or work MEM.
  • a PROC processor cooperating with a memory block BM having a memory storage and / or work MEM.
  • Such a device comprises an input module E adapted to receive a decoded or extracted excitation audio signal in a first so-called low band frequency band ( u ( n ) or U ( k )) and the parameters of a linear prediction synthesis filter ( ⁇ ( z )). It comprises an output module S adapted to transmit the synthesized and optimized high frequency signal (u HB '(n)) for example to a filtering module such as block 710 of FIG. figure 7 or a resampling module like module 311 of the figure 3 .
  • a filtering module such as block 710 of FIG. figure 7 or a resampling module like module 311 of the figure 3 .
  • the memory block may advantageously comprise a computer program comprising code instructions for carrying out the steps of the method for determining an optimized scale factor to be applied to an excitation signal or to a filter within the meaning of FIG. invention, when these instructions are executed by the processor PROC, and in particular the steps of determination (E602) of a linear prediction filter called additional filter, of order less than the linear prediction filter of the first frequency band, the coefficients additional filter being obtained from the parameters decoded or extracted from the first frequency band, calculation (E603) of an optimized scale factor according to at least the coefficients of the additional filter.
  • the description of the figure 6 takes the steps of an algorithm of such a computer program.
  • the computer program can also be stored on a memory medium readable by a reader of the device or downloadable in the memory space thereof.
  • the memory MEM generally records all the data necessary for the implementation of the method.
  • the device thus described may also include the functions of applying the optimized scaling factor to the extended excitation signal, frequency band extension, low band decoding and other processing functions. described for example in figure 3 and 4 in addition to the optimized scale factor determination functions according to the invention.

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EP14749907.3A 2013-07-12 2014-07-04 Facteur d'échelle optimisé pour l'extension de bande de fréquence dans un décodeur de signaux audiofréquences Active EP3020043B1 (fr)

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Application Number Priority Date Filing Date Title
FR1356909A FR3008533A1 (fr) 2013-07-12 2013-07-12 Facteur d'echelle optimise pour l'extension de bande de frequence dans un decodeur de signaux audiofrequences
PCT/FR2014/051720 WO2015004373A1 (fr) 2013-07-12 2014-07-04 Facteur d'échelle optimisé pour l'extension de bande de fréquence dans un décodeur de signaux audiofréquences

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