EP2347411B1 - Vor-echo-dämpfung in einem digitalaudiosignal - Google Patents

Vor-echo-dämpfung in einem digitalaudiosignal Download PDF

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Publication number
EP2347411B1
EP2347411B1 EP09747881A EP09747881A EP2347411B1 EP 2347411 B1 EP2347411 B1 EP 2347411B1 EP 09747881 A EP09747881 A EP 09747881A EP 09747881 A EP09747881 A EP 09747881A EP 2347411 B1 EP2347411 B1 EP 2347411B1
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sub
signal
attenuation
block
blocks
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French (fr)
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EP2347411A1 (de
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Balazs Kovesi
Stéphane RAGOT
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Orange SA
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France Telecom 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
    • G10L19/26—Pre-filtering or post-filtering
    • 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
    • G10L19/0212—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 using orthogonal transformation
    • 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
    • G10L19/022—Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
    • G10L19/025—Detection of transients or attacks for time/frequency resolution switching

Definitions

  • the invention relates to a method and a device for attenuating pre-echoes when decoding a digital audio signal.
  • compression processes for the transport of digital audio signals on the transmission networks, whether for example fixed or mobile networks, or for the storage of signals, compression processes (or source coding) using coding systems of the time coding type or frequency coding by transform.
  • the method and the device which are the subject of the invention, thus have as their field of application the compression of sound signals, in particular frequency-coded digital audio signals.
  • the figure 1 illustrates, by way of illustration, a schematic diagram of the coding and decoding of a digital audio signal by transform including an addition / overlap synthesis analysis according to the prior art.
  • Certain musical sequences such as percussion and certain segments of speech like the plosives (/ k /, / t /, ...), are characterized by extremely sudden attacks which result in very fast transitions and a very strong variation signal dynamics in a few samples.
  • An example of a transition is given to the figure 1 from sample 410.
  • the input signal is cut into blocks of samples of length L (here represented by dotted vertical lines).
  • the input signal is noted x ( n ).
  • L 160 samples.
  • the division in blocks, also called frames, operated by the transform coding is totally independent of the sound signal and the transitions appear at any point in the analysis window. But after transform decoding, the reconstructed signal is tainted by "noise” (or distortion) generated by the quantization (Q) -quantization inverse (Q -1 ) operation.
  • This coding noise is temporally distributed in a relatively uniform manner over the entire temporal support of the transformed block, that is to say over the entire length of the window of length 2L of samples (with overlap of L samples).
  • the energy of the coding noise is generally proportional to the energy of the block and is a function of the decoding rate.
  • the level of the coding noise is lower than that of the signal for the high energy samples that immediately follow the transition, but the level is higher than that of the signal for the lower energy samples, especially on the part preceding the transition (samples 160 - 410 of the figure 1 ).
  • the signal to noise ratio is negative and the resulting degradation, can appear very troublesome to listen.
  • Pre-echo is the coding noise prior to the transition and post-echo the noise after the transition.
  • the human ear also performs a post-masking of a longer duration, from 5 to 60 milliseconds, during the passage of high energy sequences to low energy sequences.
  • the rate or level of inconvenience acceptable for post-echoes is therefore greater than for pre-echoes.
  • a first solution is to apply adaptive filtering.
  • the reconstituted signal consists in fact of the original signal and the quantization noise superimposed on the signal.
  • the aforementioned filtering process does not allow to find the original signal, but provides a strong reduction of pre-echoes. However, it requires to transmit additional auxiliary parameters to the decoder.
  • a technique that does not require the transmission of auxiliary parameters is described in the French patent application FR 06 01466 .
  • the described method allows discriminating the presence of pre-echoes and attenuating the pre-echoes of a hierarchically coded digital audio signal (generating a multilayer binary stream) from a transform coding, generating pre-echo, and a time coding, not generating pre-echoes.
  • This patent application more specifically describes the decoder detection of a low energy zone preceding a transition to a high energy zone, the attenuation of the pre-echoes in the detected low energy zones and the inhibition of the attenuation.
  • pre-echoes in the area of high energy The treatment for attenuating the pre-echoes is based on a comparison between the signal resulting from a decoding by transform (generating pre-echoes) and a signal resulting from a temporal decoding (non-echo generator).
  • This technique does not require specific auxiliary information transmission from the coder but requires the presence of a reference signal from a time decoding.
  • All decoders using transform decoding do not necessarily have a reference signal from time decoding. Moreover, in the case where such a reference signal is available at the decoder, it is not always suitable for calculating the attenuation of pre-echoes.
  • a stereo scalable encoder for example the stereo extension of ITU-T G.729.1, can operate as described below.
  • the encoder calculates the average of the two left and right channels of the stereo signal, then codes this average by the G.729.1 encoder, and finally transmits additional parameters of stereo extension.
  • the bitstream transmitted to the decoder thus includes a G.729.1 layer with additional layers of stereo extension.
  • an additional first layer has parameters reflecting the difference in energy per subband (in the transformed domain) between the two channels of the stereo signal.
  • a second layer comprises, for example, the transformed coefficients of the residual signal, defined as the difference between the original signal and the signal decoded from the G.729.1 bit stream and the first layer.
  • the G.729.1 decoder in extended mode first decodes the mono signal and finds, according to the transmitted parameters, the transformed coefficients of the two left and right channels.
  • the decoding of the mono signal by a G.729.1 decoder provides a reference signal based on the average of the two channels. In the case where the difference in levels between the two channels is large, the time envelope of the mono signal will then be small relative to the output of the inverse transform of the higher level channel and strong compared to the output of the transform. inverse of the lower level channel.
  • the attenuation factor is defined on characteristics specific to the decoded signal that do not require transmission of information from the encoder or signal derived from non-echo-generating decoding.
  • a factor adapted to each sub-block of the current frame and calculated from the reconstructed signal makes it possible to improve the quality of the pre-echo attenuation processing.
  • the concatenated signal can be defined from the reconstructed signal of the current frame and the second part of the current frame as defined later with reference to the figure 2 .
  • the method does not introduce time delay.
  • the concatenated signal is defined as the reconstructed signal of the current frame and the next frame.
  • the concatenated signal can be physically stored at different locations by sub-blocks.
  • a minimum value is set for an attenuation value of the factor as a function of the temporal envelope of the reconstructed signal of the previous frame.
  • the time envelope of the reconstructed signal of the preceding frame can for example be determined by calculating the minimum energy per sub-block or by calculating the average energy or any other calculation.
  • the attenuation factor is determined according to the temporal envelope of said sub-block, the maximum of the temporal envelope of the sub-block comprising said transition and the temporal envelope. of the reconstructed signal of the previous frame.
  • the time envelope is determined by a calculation of energy by sub-blocks.
  • the method further comprises a step of calculating and storing the temporal envelope of the current frame after the attenuation step in the determined sub-blocks.
  • This time envelope calculation will therefore be used to process the next frame. This calculation is precise since the signal is no longer disturbed by the pre-echoes.
  • an attenuation factor of value 1 is assigned to the samples of said sub-block comprising the transition as well as to the samples of the following sub-blocks in the current frame.
  • Attenuation is therefore inhibited in these sub-blocks which do not include pre-echoes.
  • This particular embodiment has proved particularly effective and is simple to implement.
  • the method provides for the determination of a smoothing function between the calculated factors sample by sample.
  • a factor correction is performed for the sub-block preceding the sub-block having a transition, by applying an attenuation inhibiting attenuation value, to the attenuation factor applied to a predetermined number. samples of the sub-block preceding the sub-block having a transition.
  • the invention relates to a decoder of a digital audio signal comprising a device as described above.
  • Such a decoder can for example be a G.729.1-SWB / stereo decoder studied in Question 23 of ITU-T, Commission 16.
  • the invention can be integrated in such a decoder in stereo mode or in SWB mode (for "super wide band” English).
  • the invention is directed to a computer program comprising code instructions for implementing the steps of the attenuation method as described, when these instructions are executed by a processor.
  • the figure 2 represents a frame of the decoded signal as well as the configuration of the reconstructed recovery signal as described with reference to FIG. figure 1 .
  • the intermediate signal comprises an antisymmetrical portion and a symmetrical portion.
  • the pre-echo attenuation method generates a concatenated signal [x rec, N (0). X ,,,,, N (L-1) x cur2h, N (0) ... x cur2h, N (L-1)], from the reconstructed signal of the current frame x rec, N (n) and the signal of the second part of the current frame upgraded x cur2h, N (n).
  • This concatenated signal is divided into sub-blocks of samples of determined length, here an even number.
  • the method determines the sub-blocks of the current block requiring attenuation of pre-echoes.
  • the attenuation method also includes a step of calculating the attenuation factor to be applied to the determined sub-blocks. The calculation is performed for each of the sub-blocks as a function of the temporal envelope of the concatenated signal.
  • This calculation can also be performed in addition to the time envelope of the reconstructed signal of the previous frame.
  • An attenuation device 100 includes a module 101 for defining a concatenated signal, a module 102 for dividing the concatenated signal into sub-blocks, a module 103 for calculating temporal envelope of the concatenated signal, a detection unit 104 transition from the time envelope to a high energy area and determination of the low energy sub-blocks preceding a sub-block in which a transition has been detected and an attenuation module 105 in the determined sub-blocks.
  • the attenuation module is able to apply an attenuation factor to the sub-blocks determined by the module 104, the attenuation factor being determined by the attenuation module as a function of the time envelope of the concatenated signal.
  • the attenuation device is included in a decoder comprising an inverse quantization module 110 (Q -1 ), an inverse transform module 120 (MDCT -1 ), an add / overlap signal reconstruction module 130 (add / rec) as described with reference to figure 1 and delivering a reconstructed signal to the attenuation device according to the invention.
  • the Figures 4a and 4b illustrate examples of signals with transitions or attacks in the signal.
  • the pre-echo phenomenon exists when the energy of a part of the signal in an MDCT window is significantly higher (attack) than that of the other parts.
  • the pre-echo is then observed in the low energy parts before the attack. It is therefore in this part that the pre-echoes must be attenuated.
  • attack or the transition of the signal is in the current frame (L first samples) or in the next frame (L following samples) corresponding to the second part of the current frame as represented in figure 2 .
  • the figure 4a represents a signal concatenated with an attack of the signal in the second part of the current frame.
  • the second part of the current frame is symmetric by property of the inverse transform MDCT. Indeed according to the invention the pre-echoes are attenuated without introducing additional delay in the transform decoding.
  • the pre-echo attenuation method according to the invention delivers pre-echo attenuation factors for each sample of the frame. This process will now be described with reference to figures 5 and 6 .
  • the organization chart represented in figure 5 illustrates the different stages of calculation of the attenuation factor according to the invention for a current frame.
  • step 201 the time envelope of the reconstructed signal of the current frame is calculated and in step 202, the temporal envelope of the second part of the updated current frame is calculated.
  • the temporal envelope is for example obtained by calculating the energy by sub-blocks as described with reference to the figure 6 . It can be obtained by other methods, for example by calculating the average of the absolute values of the signal by sub-blocks, or else the maximum value or the median value of each sub-block.
  • the envelope can also be obtained for example as a Teager-Kaiser type operator followed by a low-pass filtering. In any case, it is assumed here, without loss of generality, that the temporal envelope is defined with a temporal resolution of one value per sub-block, the size of the sub-blocks being flexible.
  • an attenuation factor function is defined from the envelopes of the current frame defined in steps 201 and 202 and from the envelope of the reconstructed signal of the previous frame (T env (x rec, N-1 (n)).
  • the optional step 204 defines a smoothing function on the obtained values of the attenuation factor in order to avoid discontinuities that could be revealed in the processed signal.
  • step 302 the energy En (k) of the K 2 sub-blocks of the reconstructed signal x rec, N (n) is calculated.
  • step 303 the energy of each sub-block of the second part of the current frame being upgraded x cur2h, N (n), is calculated. Only K 2/2 values are different because of the symmetry of this part of the signal as represented in figure 4a .
  • step 305 a loop counter is initialized.
  • the sub-block of index ind1 is determined at 307 to have an attenuation factor g (k) as a function of its energy En (k). max maximum energy in and of the average energy of the reconstructed signal of the previous frame x rec, N-1 and this factor is assigned in 308 to all the samples of the sub-block.
  • step 310 the index of the first sample of the sub-block is calculated at maximum energy.
  • step 311 it is checked whether it is less than the length of the frame. If so, the maximum energy sub-block is in the current frame and the factor 1, ie, a value inhibiting attenuation, is assigned to all samples from the beginning of the sub-block to the current frame. at the end of the frame in the loop of steps 311-312-313.
  • step 314 the average energy of the reconstructed current frame, that is to say the first K 2 blocks of the reconstructed signal x rec, N ( n ), is calculated and stored. It will be used in the following frame for the calculation of the new factors.
  • a factor smoothing function is determined and applied sample by sample to avoid abrupt factor variations.
  • the last attenuation factor obtained for the last sub-block to be attenuated of the current frame is stored for use in the next frame in step 315.
  • smoothing functions are possible, such as, for example, a linear transition between the two factor values, either with a constant slope (for example in steps of 0.05) or with a fixed length (for example, on 16 samples).
  • the step 307 for calculating the attenuation factor for a sub-block is now detailed in a particular embodiment of the invention with reference to FIG. figure 7 .
  • the ratio max / en (k) of the maximum energy determined in step 304 on the energy of the sub-block treated is first calculated in step 401.
  • S1 is fixed at 16 in the example, this value being optimized experimentally.
  • the factor is then set at step 403 at an attenuation inhibiting attenuation value, i.e. 1.
  • step 404 it is tested in step 404 if the ratio r is less than or equal to a second threshold S2.
  • S2 is set at 32 in the example, this value being optimized experimentally.
  • the risk of pre-echo is then maximum and is applied in step 406 a strong attenuation value to the factor, for example 0.1.
  • the frame that precedes the pre-echo frame has a homogeneous energy that corresponds to the energy of the background noise at that time. According to the experience it is not useful or even desirable that the signal energy becomes lower than the average energy of the previous frame after pre-echo processing.
  • a limit value of the lim r factor r is calculated with which the given sub-block is obtained exactly the same energy as the average energy of the previous frame. Then at step 408, this value is limited to a maximum of 1 since the attenuation values are of interest here.
  • the thus obtained lim value g serves as the lower limit in the final calculation of the attenuation factor at step 409.
  • a rate characteristic of the transmitted signal may be taken into account. Indeed, in a low-rate transmission, the quantization noise is generally important, which increases the risk of annoying pre-echo. In contrast, at very high speed, the coding quality can be very good and no pre-echo attenuation is necessary.
  • the rate information can therefore be taken into account in determining the attenuation factor.
  • the figures 8a and 8b illustrate the implementation of the attenuation method of the invention in a typical example.
  • the signal is sampled at 8 kHz, the frame length is 160 samples and each frame is divided into 4 sub-blocks of 40 samples.
  • part b) of the figure 8a the result of decoding (only the left channel) without pre-echo processing is shown.
  • Part c. Shows the evolution of the pre-echo attenuation factor (continuous line) obtained by implementing the method according to the invention.
  • the dotted line represents the factor before smoothing.
  • Part d. Illustrates the result of the decoding after application of the pre-echo processing (multiplication of the signal b.) With the signal c.)). We see that the pre-echo has been removed.
  • the figure 8b illustrates the same typical example for which an implementation of an alternative embodiment of the attenuation method according to the invention is carried out.
  • Part c. figure 8b give an example of such a correction.
  • the value of factor 1 has been assigned to the last 16 samples of the sub-block preceding the sub-block with the attack, starting from index 344.
  • the smoothing function gradually increases the factor to have a value close to 1 at the time of the attack.
  • the amplitude of the attack is then preserved.
  • the difficulty of this method is to know, in the frame that precedes the frame including the attack, whether the attack is in the first sub-block or not.
  • the factor 1 value must be assigned to the last samples of the frame.
  • the problem is that on the concatenated signal the position of the attack can not be determined with certainty due to the symmetry of this part of the concatenated signal which in fact reflects the well-known "time folding" property of the MDCT transform .
  • the figures 9 and 10 illustrate the concatenated signal corresponding to the second frame of figures 8a and 8b .
  • One solution is to always assign the factor value to 1 to the last samples of the frame if the attack is detected in the 4 th sub-block of the concatenated signal. If in the next frame, the attack is in the first sub-block (case of the figure 11 ) the operation is then optimal. On the other hand, when the attack is in the 4 th sub-block (case of the figure 12 ), the attenuation is suboptimal because around the end of the frame, the pre-echo attenuation factor increases to 1 for a few samples and then drops back to the correct attenuation level at the beginning of the next frame. The subjective impact of this sub-optimality is low because when the attack is in the 4 th sub-block of the following frame its amplitude is well diminished by the analysis windowing. The pre-echo caused by this attack is weak.
  • the method of the invention uses a particular example of calculating the beginning of the attack (search for the maximum energy per sub-block) but can work with any other method of determining the beginning of the attack.
  • the method which is the subject of the invention mentioned above applies to the attenuation of the pre-echoes in any transform coder which uses an MDCT filter bank or any real or complex value perfect reconstruction filter bank, or the banks of almost perfect reconstruction filters as well as filter banks using the Fourier transform or the wavelet transform.
  • the pre-echo reduction method then applies directly to the reconstructed signal and no longer to the concatenated signal which is hybridized between reconstructed signal / intermediate signal with temporal folding.
  • the transition detection, attenuation factor calculation and pre-echo reduction means described above apply.
  • the concatenated signal is not defined explicitly, it is always possible to use the reconstructed signal at the current frame and an intermediate signal of the inverse MDCT to perform the operations described above.
  • FIG. 13a An example of a stereo signal encoder is described with reference to the figure 13a .
  • a suitable decoder comprising an attenuation device according to the invention is described with reference to the figure 13b .
  • the figure 13a shows an example of an encoder, for which stereo information is transmitted by frequency band and is decoded in the frequency domain.
  • a mono signal M is calculated from the input signals of the left channel L and the right channel R by die-stamping means 500.
  • the encoder also integrates time-frequency transformation means 502, 503 and 504 capable of producing a transform, for example a discrete Fourier Transform or DFT (of the English "Discrete Fourier Transform"), an MDCT transform (of the “Modified Discrete Cosine Transform”), an MCLT (Modulated Complex Lapped Transform).
  • a discrete Fourier Transform or DFT of the English "Discrete Fourier Transform”
  • MDCT transform of the "Modified Discrete Cosine Transform”
  • MCLT Modulated Complex Lapped Transform
  • the mono signal M is also quantized and coded by the means 501, for example by the G.729.1 coder standardized in ITU-T.
  • This module delivers the binary bit stream bst 1 and also the decoded mono signal M transformed in the frequency domain.
  • the module 505 performs the stereo parametric coding from the frequency signals L, R, and M and the decoded signal M. It delivers the first optional extension layer of the bit stream bst 2 and the two channels of the decoded stereo signal L and R obtained by decoding the two layers bst 1 and bst 2 .
  • the stereo residual signal in the frequency domain is calculated by the means 506 and 507 and encoded by the coding means 508 and the second optional extension layer of the bitstream bst 3 is obtained.
  • the encoded heart signal bst 1 and the optional extension layers bst 2 and bst 3 are transmitted to the decoder.
  • the figure 13b shows an example of a decoder capable of receiving the encoded heart signal bst 1 and the optional extension layers bst 2 and bst 3 .
  • Decoding means 600 make it possible to decode the binary bit stream bst 1 and to obtain the decoded mono signal M. If the first optional extension layer bst 2 is available it can be decoded by the parametric stereo decoding means 601 to build the decoded stereo signal L and R from the mono decoded signal M. Otherwise, L and R will be equal to M.
  • the second optional extension layer bst 3 When the second optional extension layer bst 3 is also available it is decoded by the decoding means 602 to obtain the stereo residual signal in the frequency domain. This is in addition to the decoded stereo signal L and R to increase the accuracy of the frequency representation of the signal. Otherwise, when this second extension layer is not available L and R remain unchanged.
  • the figure 14a shows an exemplary encoder of the super-wide band extension of a G.729.1 type wideband encoder.
  • the super-wideband input signal S 32 is downsampled by the subsampling means 700 to obtain an expanded band signal S 16 .
  • This signal is quantized and coded by the means 701 for example by the ITU G.729.1 coder.
  • This module delivers the binary bit stream bst 1 and also the decoded broadband signal ⁇ 16 in the frequency domain.
  • the super-wideband input signal S 32 is transformed in the frequency domain by the transformation means 704.
  • the frequencies of the high band (7000 -14000Hz band) that are not coded in the enlarged band portion will be encoded by the means of transmission. This coding is based on the spectrum of the decoded broadband signal ⁇ 16 .
  • the coded parameters constitute the first optional extension of the bst 2 binary train.
  • An optional second layer of the bit stream bst 3 provided by the coding means 705 contains the parameters for improving the quality of the enlarged band (50-7000 Hz).
  • the decoder of the figure 14b represents a super-wideband decoder (50-14000 Hz) corresponding to the encoder of the figure 14a .
  • the binary bit stream bst 1 is decoded by a G.729.1 type wideband encoder (module 800).
  • the spectrum of the broadband decoded signal is thus obtained. This spectrum is possibly improved by the decoding at 801 of the second extension layer optional bst 3 .
  • the module 801 also includes the frequency-time transformation of the broadband signal.
  • the present invention does not intervene in this frequency-time transformation to reduce the pre-echoes because here we have the echo-free time signals (CELP and TDBWE components of the G.729.1 coder) and therefore the technique described in the French patent application. FR 06 01466 can be applied.
  • the decoded broadband signal is then oversampled by a factor of 2 in the oversampling means 802.
  • the first optional extension layer bst 2 When the first optional extension layer bst 2 is available at the decoder, it is decoded by the decoding means 803.
  • This decoding is based on the spectrum of the decoded broadband signal ⁇ 16 .
  • the spectrum thus obtained contains the non-zero values only in the 7000-14000 Hz frequency zone not coded by the enlarged band part. In this configuration, between 7000 and 14000 Hz, therefore, there are no reference signals without pre-echo.
  • the attenuation device according to the invention is therefore implemented.
  • the time signal is obtained by frequency-time inverse transformation by the module 504.
  • the addition / recovery reconstruction module provides a reconstructed signal.
  • the reduction of the pre-echoes according to the present invention is carried out by the attenuation module 807 as described with reference to FIG. figure 3 .
  • the signal after inverse transformation MDCT contains only frequencies higher than 7000 Hz.
  • the temporal envelope of this signal can therefore be determined with a very high precision, which increases the efficiency of the attenuation. pre-echoes by the attenuation method of the invention.
  • this device 100 in the sense of the invention typically comprises a ⁇ P processor cooperating with a memory block BM including a storage and / or working memory, and a memory buffer MEM mentioned above as a means for storing, for example the time envelope of the frame current attenuation factor calculated for the last sample of the current frame, the energy of the sub-blocks of the current frame or any other data necessary for the implementation of the attenuation method as described with reference to the Figures 5 to 7 .
  • This device receives as input successive frames of the digital signal Se and delivers the reconstructed signal Sa with attenuation of pre-echoes if necessary.
  • the memory block BM may comprise a computer program comprising the code instructions for implementing the steps of the method according to the invention when these instructions are executed by a ⁇ P processor of the device and in particular a step of defining a concatenated signal, from at least the reconstructed signal of the current frame, a step of dividing said concatenated signal into sub-blocks of samples of a determined length, a time envelope calculation step of the concatenated signal, a transition detection step of the temporal envelope to a high energy area, a step of determining the low energy sub-blocks preceding a sub-block in which a transition has been detected and an attenuation step in the determined sub-blocks.
  • the attenuation is performed according to an attenuation factor calculated for each of the determined sub-blocks, as a function of the temporal envelope of the concatenated signal.
  • the Figures 5 to 7 can illustrate the algorithm of such a computer program.
  • This attenuation device can be independent or integrated into a digital signal decoder.

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Claims (12)

  1. Verfahren zur Dämpfung von Vorechos in einem audiodigitalen Signal, das ausgehend von einer Transformationscodierung erzeugt wird, bei dem bei der Decodierung für einen laufenden Rahmen dieses audiodigitalen Signals das Verfahren aufweist:
    - einen Schritt der Definition (CONC) eines verketteten Signals ausgehend mindestens vom rekonstruierten Signal des laufenden Rahmens;
    - einen Schritt der Teilung (DIV, 301) des verketteten Signals in Teilblöcke von Tastproben bestimmter Länge;
    - einen Schritt des Berechnens (ENV, 302) einer zeitlichen Hüllkurve des verketteten Signals;
    - einen Schritt der Erfassung (DETECT, 304) des Übergangs der zeitlichen Hüllkurve in eine Zone mit starker Energie;
    - einen Schritt der Bestimmung (DETECT, 304) der Teilblöcke niedriger Energie, die vor einem Teilblock liegen, in dem ein Übergang erfasst wurde, und gekennzeichnet durch
    - einen Schritt der Dämpfung (ATT) in den bestimmten Teilblöcken,
    wobei das Verfahren dadurch gekennzeichnet ist, dass die Dämpfung gemäß einem Dämpfungsfaktor durchgeführt wird, der für jeden der bestimmten Teilblöcke berechnet wird, abhängig von der zeitlichen Hüllkurve des verketteten Signals und der zeitlichen Hüllkurve des rekonstruierten Signals des vorhergehenden Rahmens.
  2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass ein Mindestwert für einen Dämpfungswert des Faktors abhängig von der zeitlichen Hüllkurve des rekonstruierten Signals des vorhergehenden Rahmens festgelegt wird.
  3. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass der Dämpfungsfaktor abhängig von der zeitlichen Hüllkurve des Teilblocks, dem Maximum der zeitlichen Hüllkurve des den Übergang enthaltenden Teilblocks und der zeitlichen Hüllkurve des rekonstruierten Signals des vorhergehenden Rahmens bestimmt wird.
  4. Verfahren nach einem der Ansprüche 1 bis 3, dadurch gekennzeichnet, dass die zeitliche Hüllkurve durch eine Energieberechnung pro Teilblöcke bestimmt wird.
  5. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass es außerdem einen Schritt der Berechnung und der Speicherung der zeitlichen Hüllkurve des laufenden Rahmens nach dem Schritt der Dämpfung in den bestimmten Teilblöcken aufweist.
  6. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass ein Dämpfungsfaktor des Werts 1 den Tastproben des den Übergang aufweisenden Teilblocks sowie den Tastproben der folgenden Teilblöcke im laufenden Rahmen zugeteilt wird.
  7. Verfahren nach Anspruch 4, dadurch gekennzeichnet, dass der Dämpfungsfaktor pro bestimmtem Teilblock gemäß den folgenden Schritten bestimmt wird:
    - Berechnung des Verhältnisses der bestimmten maximalen Energie in dem einen Übergang aufweisenden Teilblock zur Energie des laufenden Teilblocks;
    - Vergleich des Verhältnisses mit einer ersten Schwelle;
    - wenn das Verhältnis niedriger als die oder gleich der ersten Schwelle ist, Zuteilung eines die Dämpfung verhindernden Werts zum Dämpfungsfaktor;
    - wenn das Verhältnis höher als die erste Schwelle ist:
    . Vergleich des Verhältnisses mit einer zweiten Schwelle;
    . wenn das Verhältnis niedriger als die oder gleich der zweiten Schwelle ist, Zuteilung eines schwachen Dämpfungswerts zum Dämpfungsfaktor;
    . wenn das Verhältnis höher als die zweite Schwelle ist, Zuteilung eines starken Dämpfungswerts zum Dämpfungsfaktor.
  8. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass eine Glättungsfunktion zwischen den Tastprobe für Tastprobe berechneten Faktoren bestimmt wird.
  9. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass eine Faktorkorrektur für den Teilblock durchgeführt wird, der vor dem einen Übergang aufweisenden Teilblock liegt, indem ein die Dämpfung verhindernder Dämpfungswert an den Dämpfungsfaktor angewendet wird, der an eine vorbestimmte Anzahl von Tastproben des vor dem einen Übergang aufweisenden Teilblock liegenden Teilblocks angewendet wird.
  10. Vorrichtung zur Dämpfung von Vorechos in einem audiodigitalen Signal, das ausgehend von einem Transformationscodierer erzeugt wird, bei dem die einem Decodierer zugeordnete Vorrichtung zur Verarbeitung eines laufenden Rahmens dieses audiodigitalen Signals enthält:
    - ein Modul (101) zur Definition eines verketteten Signals ausgehend mindestens von dem rekonstruierten Signals des laufenden Rahmens;
    - ein Modul (102) zur Teilung des verketteten Signals in Teilblöcke von Tastproben bestimmter Länge;
    - ein Modul (103) zur Berechnung einer zeitlichen Hüllkurve des verketteten Signals;
    - ein Modul (104) zur Erfassung des Übergangs der zeitlichen Hüllkurve in eine Zone starker Energie;
    - ein Modul (104) zur Bestimmung der Teilblöcke niedriger Energie, die vor einem Teilblock liegen, in dem ein Übergang erfasst wurde; und gekennzeichnet durch
    - ein Modul (105) zur Dämpfung in den bestimmten Teilblöcken,
    wobei die Vorrichtung dadurch gekennzeichnet ist, dass das Dämpfungsmodul die Dämpfung gemäß einem für jeden der bestimmten Teilblöcke berechneten Dämpfungsfaktor abhängig mindestens von der zeitlichen Hüllkurve des verketteten Signals und der zeitlichen Hüllkurve des rekonstruierten Signals des vorhergehenden Rahmens durchführt.
  11. Decodierer eines audiodigitalen Signals, der eine Vorrichtung nach Anspruch 10 aufweist.
  12. EDV-Programm, das Codeanweisungen für die Durchführung der Schritte des Verfahrens nach einem der Ansprüche 1 bis 9 aufweist, wenn diese Anweisungen von einem Prozessor ausgeführt werden.
EP09747881A 2008-09-17 2009-09-15 Vor-echo-dämpfung in einem digitalaudiosignal Active EP2347411B1 (de)

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FR0856248 2008-09-17
PCT/FR2009/051724 WO2010031951A1 (fr) 2008-09-17 2009-09-15 Attenuation de pre-echos dans un signal audionumerique

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BR112012009032B1 (pt) * 2009-10-20 2021-09-21 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e. V. Codificador de sinal de áudio, decodificador de sinal de áudio, método para prover uma representação codificada de um conteúdo de áudio, método para prover uma representação decodificada de um conteúdo de áudio para uso em aplicações de baixo retardamento
FR2992766A1 (fr) * 2012-06-29 2014-01-03 France Telecom Attenuation efficace de pre-echos dans un signal audionumerique
FR3000328A1 (fr) * 2012-12-21 2014-06-27 France Telecom Attenuation efficace de pre-echos dans un signal audionumerique
EP2830065A1 (de) 2013-07-22 2015-01-28 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zur Decodierung eines codierten Audiosignals unter Verwendung eines Überschneidungsfilters um eine Übergangsfrequenz
FR3025923A1 (fr) * 2014-09-12 2016-03-18 Orange Discrimination et attenuation de pre-echos dans un signal audionumerique
US10354667B2 (en) * 2017-03-22 2019-07-16 Immersion Networks, Inc. System and method for processing audio data

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FR2674710B1 (fr) * 1991-03-27 1994-11-04 France Telecom Procede et systeme de traitement des preechos d'un signal audio-numerique code par transformee frequentielle.
DE19736669C1 (de) * 1997-08-22 1998-10-22 Fraunhofer Ges Forschung Verfahren und Vorrichtung zum Erfassen eines Anschlags in einem zeitdiskreten Audiosignal sowie Vorrichtung und Verfahren zum Codieren eines Audiosignals
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DE10324438A1 (de) * 2003-05-28 2004-12-16 Knorr-Bremse Systeme für Schienenfahrzeuge GmbH Bremsvorrichtung eines Schienenfahrzeugs
SE527670C2 (sv) * 2003-12-19 2006-05-09 Ericsson Telefon Ab L M Naturtrogenhetsoptimerad kodning med variabel ramlängd
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CN102160114A (zh) 2011-08-17
RU2011115003A (ru) 2012-10-27
KR101655913B1 (ko) 2016-09-08
KR20110076936A (ko) 2011-07-06
US20110178617A1 (en) 2011-07-21
JP2012503214A (ja) 2012-02-02
ES2400987T3 (es) 2013-04-16
EP2347411A1 (de) 2011-07-27
US8676365B2 (en) 2014-03-18
WO2010031951A1 (fr) 2010-03-25

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