EP2867893B1 - Wirksame prä-echodämpfung in einem digitalen audiosignal - Google Patents

Wirksame prä-echodämpfung in einem digitalen audiosignal Download PDF

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EP2867893B1
EP2867893B1 EP13744654.8A EP13744654A EP2867893B1 EP 2867893 B1 EP2867893 B1 EP 2867893B1 EP 13744654 A EP13744654 A EP 13744654A EP 2867893 B1 EP2867893 B1 EP 2867893B1
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Prior art keywords
echo
attack
signal
filtering
attenuation
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French (fr)
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EP2867893A1 (de
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Balazs Kovesi
Stéphane RAGOT
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Orange SA
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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
    • G10L19/02—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
    • G10L19/03—Spectral prediction for preventing pre-echo; Temporary noise shaping [TNS], e.g. in MPEG2 or MPEG4
    • 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
    • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
    • G10L21/0364—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude for improving intelligibility
    • 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
    • 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
    • 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 pre-echo attenuation processing method and device for 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 represents for illustrative purposes, a block 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, represented on the figure 1 by dotted vertical lines.
  • the input signal is denoted x ( n ), where n is the index of the sample.
  • 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 can therefore 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 relatively uniformly over the entire temporal support of the transformed block, that is to say on any the 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 coding / decoding rate.
  • the level of the coding noise is typically lower than that of the signal for the high energy segments that immediately follow the transition, but the level is higher than that of the signal for the lower energy segments, 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.
  • MPEG AAC Advanced Audio Coding
  • MPEG AAC Advanced Audio Coding
  • Transform encoders used for conversational applications such as ITU-T G.722.1, G.722.1C or G.719 often use a window of 40 ms duration at 16, 32 or 48 kHz (respectively) and a frame length of 20 ms . It should be noted that the ITU-T G.719 encoder incorporates a window switch mechanism with transient detection, however the pre-echo is not completely reduced at low bit rate (typically at 32 kbit / s).
  • the aforementioned filtering process does not allow to find the original signal, but provides a strong reduction of pre-echoes. However, it requires transmitting the additional parameters to the decoder.
  • Other definitions of the factor g ( k ) are possible, for example as a function of the energy En ( k ) in the current sub-block and the energy En ( k -1) in the preceding sub-block.
  • the factor g ( k ) is then set to an attenuation-inhibiting attenuation value, i.e. 1. Otherwise, the attenuation factor is between 0 and 1.
  • the frame that precedes the pre-echo frame has a homogeneous energy that corresponds to the energy of a low energy segment (typically a background noise).
  • a background noise typically a background noise
  • the attenuation factors (or gains) g ( k ) determined by sub-blocks are then smoothed by an applied smoothing function sample by sample to avoid abrupt changes in the attenuation factor at the block boundaries.
  • L ' represents the length of a sub-block.
  • g pre (-1) is the last attenuation factor obtained for the last sample of the previous sub-block
  • the signal is sampled at 32 kHz
  • part a) of the figure 2 a frame of an original signal sampled at 32 kHz is shown.
  • An attack (or transition) in the signal is located in the sub-block beginning at the index 320.
  • This signal has been coded by a low rate (24 kbit / s) MDCT type transform coder.
  • part (b) of the figure 2 the result of the decoding without pre-echo processing is illustrated.
  • the pre-echo can be observed from sample 160, in the preceding sub-blocks the one containing the attack.
  • Part c) shows the evolution of the pre-echo attenuation factor (solid line) obtained by the method described in the aforementioned prior art patent application.
  • the dashed line represents the factor before smoothing. Note here that the position of the attack is estimated around the sample 380 (in the block delimited by the samples 320 and 400).
  • the value of factor 1 has been assigned to the last 16 samples of the sub-block preceding the attack, starting from the index 364.
  • the smoothing function progressively increases the factor to have a value close to 1 at the moment. of the attack.
  • the amplitude of the attack is then preserved, as illustrated in part d) of the figure 3 however, some pre-echo samples are not attenuated.
  • Attenuation pre-echo reduction does not reduce the pre-echo to the attack level because of gain smoothing.
  • FIG 4 Another example with the same setting as that of the figure 3 is illustrated on the figure 4 .
  • This figure represents 2 frames to better show the nature of the signal before the attack.
  • the energy of the original signal before the attack is stronger (part a) than in the case illustrated by the figure 3 , and the signal before the attack is audible (samples 0 - 850).
  • part b) we can observe the pre-echo on the decoded signal without pre-echo processing in the 700-850 area.
  • the signal energy of the pre-echo zone is attenuated to the average energy of the signal preceding the treatment zone.
  • part c) the attenuation factor calculated taking into account the energy limitation is close to 1 and that the pre-echo is always present on part d) after application of the pre-echo treatment ( multiplication of the signal b) with the signal c)), despite the good leveling of the signal in the pre-echo zone.
  • the pre-echo treatment multiplication of the signal b) with the signal c)
  • the present invention improves the state of the art.
  • the spectral shaping applied improves the pre-echo attenuation.
  • the treatment makes it possible to attenuate the pre-echo components that could remain during the implementation of the pre-echo attenuation as described in the state of the art.
  • the filtering being applied to the detected position of the attack, it makes it possible to process the attenuation of the pre-echo up to the nearest attack. This therefore offsets the disadvantage of temporal attenuation echo control which is limited to a zone not going to the attack position (margin of 16 samples for example).
  • This pre-echo attenuation processing technique can be implemented with or without knowledge of a signal derived from a time decoding and for coding a monophonic signal or a stereophonic signal.
  • the adaptation of the filtering makes it possible to adapt to the signal and to remove only the disturbing parasitic components.
  • the method further comprises calculating at least one decision parameter on the filtering to be applied to the pre-echo zone and the adaptation of the filtering coefficients according to said at least one parameter of decision.
  • the treatment is then applied only when necessary to a suitable level of filtering.
  • the at least one decision parameter is a measure of the strength of the detected attack.
  • the force of the attack determines the presence of audible high-frequency components in the pre-echo zone.
  • the attack is abrupt, the risk of having an annoying parasitic component in the pre-echo zone is large and the filtering to be carried out according to the invention is then to be expected.
  • This calculation is of less complexity and makes it possible to define well the strength of the detected attack.
  • the said at least one decision parameter may also be the value of the attenuation factor in the sub-block preceding that containing the position of the attack.
  • said at least one decision parameter is based on a spectral distribution analysis of the signal of the pre-echo zone and / or of the signal preceding the pre-echo zone.
  • the adaptation of the filtering coefficients can be done in a discrete manner as a function of the comparison of at least one decision parameter with a predetermined threshold.
  • the filter coefficients can take predetermined values according to a set of values.
  • the smallest set of values being one where only two values are possible, it is a saying for example the choice between a filtering and no filtering.
  • the adaptation of the filtering coefficients is carried out continuously according to said at least one decision parameter.
  • the attenuation step is performed at the same time as the spectral shaping filtering by integrating the attenuation factors with the coefficients defining the filtering.
  • the invention relates to a decoder of a digital audio signal comprising a device as described above.
  • the invention is directed to a computer program comprising code instructions for implementing the steps of the attenuation processing method 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 processing device, optionally removable, storing a computer program implementing a method of treatment as described above.
  • a pre-echo attenuation processing device 600 is described.
  • this device implements a pre-echo attenuation method in the decoded signal, for example that described in the patent application.
  • FR 08 56248 It furthermore implements spectral shaping filtering of the pre-echo zone.
  • the device 600 comprises a detection module 601 able to implement a step of detecting (Detect.) The position of an attack in a decoded audio signal.
  • An onset is a rapid transition and a sudden change in the dynamics (or amplitude) of the signal.
  • This type of signal may be referred to by the more general term "transient”.
  • transient we will use only the terms of attack or transition to designate also transients.
  • the synthesis window MDCT contains only 415 non-zero samples, unlike the 640 samples in the case of using conventional sinusoidal windows.
  • other analysis / synthesis windows may be used, or switches between long and short windows may be used.
  • the MDCT x MDCT ( n ) memory is used which gives a time-folding version ("folding") of the future signal.
  • the figure 1 shows that the pre-echo influences the frame preceding the one where the attack is located, and it is desirable to detect an attack in the future frame which is partially contained in the MDCT memory.
  • the signal contained in the MDCT memory includes time folding (which is compensated when the next frame is received).
  • the MDCT memory serves essentially to estimate the energy by sub-blocks of the signal in the next (future) frame, and it is considered that this estimate is sufficiently precise for the purposes of the detection and reduction of pre- echo when performed with the available MDCT memory at the current frame instead of the fully decoded signal at the future frame.
  • the current frame and the MDCT memory can be seen as concatenated signals forming a signal of length (K + K ') L' cut into (K + K ') consecutive sub-blocks.
  • the ' - 1 x rec not 2 , k 0 ... , K - 1 when the k-th sub-block is in the current frame and, like:
  • the ' - 1 x MDCT not 2 , k K , ...
  • Other pre-echo detection criteria are possible without changing the nature of the invention.
  • the device 600 also comprises a determination module 602 implementing a determination step (ZPE) of a pre-echo zone preceding the detected attack position.
  • ZPE determination step
  • the energies En ( k ) are concatenated in chronological order, with the time envelope of the decoded signal first, then the envelope of the signal of the next frame estimated from the memory of the MDCT transform. According to this envelope concatenated temporal and mean energies In and In' of the previous frame, the presence of pre-echo is detected if the ratio R ( k ) is sufficiently strong.
  • the pre-echo zone does not necessarily start at the beginning of the frame, and may involve an estimate of the length of the pre-echo. If window switching is used, the pre-echo zone must be defined to take into account the windows used.
  • a module 603 of the device 600 implements a sub-block attenuation factor calculation step of the determined pre-echo area, depending on the frame in which the attack was detected and the previous frame.
  • Other definitions of the factor g ( k ) are possible, for example as a function of En ( k ) and En ( k -1).
  • the factor is then set to an attenuation-inhibiting attenuation value, i.e. 1. Otherwise, the attenuation factor is between 0 and 1.
  • the attenuation factors g ( k ) determined by sub-blocks are then smoothed by an applied smoothing function sample by sample to avoid abrupt changes in the attenuation factor at the boundaries of the blocks.
  • the module 604 of the device 600 of the figure 6 implements the attenuation (Att.) in the sub-blocks of the pre-echo zone, by the attenuation factors obtained.
  • the spectral shaping filter used is a linear filter. Since the gain multiplication operation is also a linear operation, their order can be inverted: you can also do the formatting filtering first. Spectrum of the pre-echo area then the pre-echo attenuation by multiplying each sample of the pre-echo area by the corresponding factor.
  • c (n) 0.05, 0.1, 0.15, 0.2 and 0.25.
  • the motivation for using this filter is its low complexity, its null phase and therefore its zero delay (possible because the processing stops before the end of the current frame) but also its frequency response which corresponds well to the desired low-pass characteristics for this filter.
  • this filter can compensate for the fact that the temporal attenuation of the pre-echo is typically limited to a zone not going to the position of the attack (with a margin of, for example, 16 samples), then that the spectral shaping filtering as defined by the transfer function c ( n ) z -1 + (1 - 2c ( n )) + c ( n ) z can be applied up to the position of the attack , possibly with some interpolation samples of the filter coefficients.
  • the proposed FIR filter makes it easy to smoothly move from the unfiltered domain to the filtered domain and vice versa, by interpolation or slow variation of its coefficients.
  • the filter c ( n ) z -1 + (1-2c ( n )) + c ( n ) z can attenuate the high frequencies before the attack without modifying the attack itself. even.
  • Part d) of the figure 8 An exemplary digital audio signal, for which the processing as described herein is performed, is illustrated in part d) of the figure 8 .
  • Parts a), b) and c) of this figure show the same signals as those described with reference to the figure 4 previously.
  • Part d) differs by the implementation of filtering according to the invention. It can thus be noted that the troublesome high-frequency component is greatly reduced, so that the signal decoded after filtering has a better quality than that described in part d) of FIG. figure 4 .
  • An order 1 filter of the form c ( n ) z -1 + (1- c ( n )) can also be used in an embodiment which does not form part of the invention.
  • the filtering implemented according to the described method is an adaptive filtering. It can thus be adapted to the characteristics of the decoded audio signal.
  • a step of calculating a decision parameter (P) on the filtering to be applied to the pre-echo zone is implemented in the calculation module 605 of FIG. figure 6 .
  • this decision parameter is representative of the presence of high frequency components in the pre-echo zone.
  • the spectral shaping filter is applied, according to the invention, from the beginning of the current frame to the position pos position of the attack.
  • the spectral shaping of the pre-echo zone by filtering according to the invention is adaptive as a function of the parameter P and the attenuation values.
  • the filtering is either applied with coefficients [0.25, 0.5, 0.25], or deactivated with coefficients [0, 1, 0].
  • the filter coefficients are then adapted in a discrete manner limited to a predefined set of values.
  • the adaptation of the filter coefficients (making it possible to adapt the attenuation level of the high frequencies) is thus determined by decision parameters which measure the force of the attack, such as the parameters P and g ( k -1).
  • a progressive transition between these two filters can be carried out using also for example the intermediate filters of coefficient [0.05, 0.9, 0.05], [0.1, 0.8, 0.1], [0.15, 0.7, 0.15] and [0.2, 0.6, 0.2 ].
  • decision parameters can also be used in the decision of the choice and the adaptation of the filter, such as for example the zero crossing rate of the decoded signal of the pre-echo zone. the current frame and / or the previous frame.
  • sgn x rec boy Wut not - 1 - sgn x rec , boy Wut not
  • or sgn x ⁇ 1 i f x ⁇ 0 - 1 i f x ⁇ 0
  • a high rate of zero crossing zc in the previous frame signals the presence of high frequencies in the signal.
  • zc > L / 2 on the previous frame it is better not to apply filtering vs not z - 1 + 1 - 2 vs not + vs not z .
  • a pre-filtering of the decoded signal is also possible before calculating the zero crossing rate, or the number of zero crossings of the estimated derivative x rec , g ( n ) -x rec , g ( n -1) can be used.
  • a spectral analysis of the signal can also be made to assist the decision.
  • the spectral envelope in the MDCT domain resulting from the MDCT coding / decoding can be exploited in the choice of the filter to be used, however this variant assumes that the analysis / synthesis windows MDCT are sufficiently short for the local statistics of the MDCT to be used. signal before the attack remain stable over the length of a window.
  • one may filter the signal in the pre-echo region and the frame passed by an additional high pass filter as - c (n) z -1 + (1-2 c (n)) - c (n) z , with for example c ( n ) 0.25, and then we will choose the value of c ( n ) so that the average energy of the filtered signals in the pre-echo zone and on the past frame are as close as possible ; the choice of c ( n ) can be made on a limited set of possible values shown in figure 7 or from the ratio of energy (or an equivalent quantity such as the square root of the energy) of the signal after high-pass filtering in the pre-echo zone and in the past frame.
  • the value of c ( n ) can be fixed according to the prediction coefficient r (1) / r (0) from a linear prediction analysis (LPC) for the order 1 of the signal in the pre-echo zone and the signal in the past frame.
  • LPC linear prediction analysis
  • the decision parameter on the filtering to be applied to the pre-echo zone is based on a spectral distribution analysis of the signal the pre-echo zone and / or the preceding signal of the pre-echo zone; if the signal preceding the pre-echo zone already contains many high frequencies or if the amount of the high frequencies of the signal in the pre-echo zone and the signal preceding the pre-echo zone is substantially identical, the filtering according to the invention is not necessary and may even cause slight degradation. In these cases, the filtering according to the invention must be deactivated or attenuated by setting c (n) to 0 or to a low value close to 0.
  • the order between the attenuation and filtering step may be reversed.
  • the spectral shaping filtering (F) is done before the attenuation (Att.).
  • the attenuation of the amplitudes can also be combined (or integrated) by defining a set of "conjoint" filter coefficients, for example if for the sample n the filter has coefficients [c ( n ), 1-2 c ( n ), c ( n )] and the attenuation factor is g ( n ), we can directly use the filter [ g pre ( n ) c ( n ), g pre ( n ) 2g pre ( n ) c ( n) ), g pre ( n ) c ( n )].
  • the figure 11 illustrates the benefit of making adaptive filtering. It uses the same signals parts a), b) and c) that the figure 10 and illustrates the fact that the implementation of the nonadaptive filtering shown in part d), unnecessarily modifies the signal in the case where the high-frequency components are already present in the signal to be encoded. It is observed that from sample 640 the high frequencies are unnecessarily attenuated, which could lead to a slight deterioration in quality.
  • the use of an adaptive filtering as described above makes it possible to inhibit or attenuate the filtering under these conditions, to not remove high frequencies already present in the signal to be coded and thus to avoid possible degradation due to the filtering.
  • the attenuation processing device 600 as described is here included in a decoder comprising a reverse quantization module 610 (Q -1 ) receiving a signal S, a reverse transformation module 620 (MDCT -1 ), a module 630 signal reconstruction by addition / recovery (add / rec) as described with reference to the figure 1 and delivering a reconstructed signal to the attenuation processing device according to the invention.
  • a decoder comprising a reverse quantization module 610 (Q -1 ) receiving a signal S, a reverse transformation module 620 (MDCT -1 ), a module 630 signal reconstruction by addition / recovery (add / rec) as described with reference to the figure 1 and delivering a reconstructed signal to the attenuation processing device according to the invention.
  • a processed signal Sa is provided in which a pre-echo attenuation has been performed.
  • the processing performed improved the pre-echo attenuation by attenuating, if necessary, the high frequency components in the pre-echo area.
  • 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 detecting a position of etching in the decoded signal, of determining a pre-echo zone preceding the detected attack position in the decoded signal, of calculating sub-block attenuation factors of the pre-echo zone, as a function of the frame in which the attack was detected and the previous frame, pre-echo attenuation in the sub-blocks of the pre-echo area by the corresponding attenuation factors and further, a a step of applying a spectral shaping filtering of the pre-echo zone on the current frame to the detected position of the attack, the filtering having a finite impulse response to a zero transfer function phase: vs not z - 1 + 1 - 2 vs not + vs not z with c (n) a coefficient between 0 and 0.25.
  • This attenuation device can be independent or integrated into a digital signal decoder.

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  • Audiology, Speech & Language Pathology (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
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  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)

Claims (12)

  1. Verfahren zum Verarbeiten der Dämpfung eines Vorechos in einem digitalen Audiosignal, das anhand einer Codierung durch Transformation erzeugt wird, wobei bei der Decodierung das Verfahren die folgenden Schritte umfasst:
    - Detektieren (601, Detect.) einer Angriffsposition in dem detektierten Signal;
    - Bestimmen (602, ZPE) einer Vorechozone, die der in dem detektierten Signal detektierten Angriffsposition vorhergeht;
    - Berechnen (603, F. Att.) von Dämpfungsfaktoren pro Unterblock der Vorechozone als Funktion wenigstens des Rahmens, in dem der Angriff detektiert worden ist, und des vorhergehenden Rahmens;
    - Dämpfen (604, Att.) des Vorechos in den Unterblöcken der Vorechozone durch die entsprechenden Dämpfungsfaktoren;
    wobei das Verfahren dadurch gekennzeichnet ist, dass es außerdem Folgendes umfasst:
    - Anwenden einer adaptiven Filterung (606, F) zum spektralen Formen der Vorechozone in dem aktuellen Rahmen bis zu der detektierten Position des Angriffs, wobei die Filterung eine endliche Impulsantwort mit Phase null der Übertragungsfunktion ist: c n z − 1 + 1 + 2 c n + c n z ,
    Figure imgb0041
    wobei c(n) ein Koeffizient im Bereich von 0 bis 0,25 ist.
  2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass es außerdem das Berechnen wenigstens eines Entscheidungsparameters für die Filterung, der auf die Vorechozone anzuwenden ist, und das Anpassen der Filterkoeffizienten als Funktion des wenigstens einen Entscheidungsparameters umfasst.
  3. Verfahren nach Anspruch 2, dadurch gekennzeichnet, dass der wenigstens eine Entscheidungsparameter ein Maß der detektierten Angriffskraft ist.
  4. Verfahren nach Anspruch 2, dadurch gekennzeichnet, dass der wenigstens eine Entscheidungsparameter der Wert des Dämpfungsfaktors in dem Unterblock ist, der jenem vorhergeht, der die Angriffsposition enthält.
  5. Verfahrene nach Anspruch 2, dadurch gekennzeichnet, dass der wenigstens eine Entscheidungsparameter auf einer Analyse der Spektrallinienverteilung des Signals der Vorechozone und/oder des Signals, das der Vorechozone vorhergeht, beruht.
  6. Verfahren nach Anspruch 3, dadurch gekennzeichnet, dass die Messung der detektierten Angriffskraft die folgende Form besitzt: P = max ( EN k , EN k + 1 / min EN k − 1 , EN k − 2 ,
    Figure imgb0042
     wobei die Nummer des Unterblocks ist, in dem der Angriff detektiert worden ist, und EN(k) die Energie des k-ten Unterblocks ist.
  7. Verfahren nach Anspruch 2, dadurch gekennzeichnet, dass die Anpassung der Filterkoeffizienten auf diskrete Weise als Funktion des Vergleichs wenigstens eines Entscheidungsparameters mit einem vorgegebenen Schwellenwert erfolgt.
  8. Verfahren nach Anspruch 2, dadurch gekennzeichnet, dass die Anpassung der Filterkoeffizienten auf kontinuierliche Weise als Funktion des wenigstens einen Entscheidungsparameters erfolgt.
  9. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass der Dämpfungsschritt gleichzeitig mit der Filterung für die spektrale Formung durch Integrieren der Dämpfungsfaktoren in die Koeffizienten, die die Filterung definieren, ausgeführt wird.
  10. Vorrichtung zum Verarbeiten der Dämpfung eines Vorechos in einem digitalen Audiosignal, das anhand einer Codierung durch Transformation erzeugt wird, wobei die Vorrichtung, die einem Decodierer zugeordnet ist, Folgendes umfasst:
    - ein Detektionsmodul (601) zum Detektieren einer Angriffsposition in dem decodierten Signal;
    - ein Bestimmungsmodul (602) zum Bestimmen einer Vorechozone, die der detektierten Angriffsposition in dem decodierten Signal vorhergeht;
    - ein Berechnungsmodul (603) für Dämpfungsfaktoren pro Unterblock der Vorechozone als Funktion wenigstens des Rahmens, in dem der Angriff detektiert worden ist, und des vorhergehenden Rahmens;
    - ein Dämpfungsmodul (604) zum Dämpfen der Vorechos in den Unterblöcken der Vorechozone durch die entsprechenden Dämpfungsfaktoren;
    wobei die Vorrichtung dadurch gekennzeichnet ist, dass sie außerdem Folgendes umfasst:
    - ein adaptives Filterungsmodul (606) zum Ausführen der spektralen Formung der Vorechozone in dem aktuellen Rahmen bis zu der detektierten Angriffsposition, wobei die Filterung eine endliche Impulsantwort mit Phase null der folgenden Übertragungsfunktion besitzt: c n z − 1 + 1 + 2 c n + c n z ,
    Figure imgb0043
    wobei c(n) ein Koeffizient im Bereich von 0 bis 0,25 ist.
  11. Decodierer für digitales Audiosignal, der eine Vorrichtung nach Anspruch 10 umfasst.
  12. Informatikprogramm, das Codebefehle enthält, um die Schritte des Verfahrens nach einem der Ansprüche 1 bis 9 auszuführen, wenn diese Befehle durch einen Prozessor ausgeführt werden.
EP13744654.8A 2012-06-29 2013-06-28 Wirksame prä-echodämpfung in einem digitalen audiosignal Active EP2867893B1 (de)

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FR1256285A FR2992766A1 (fr) 2012-06-29 2012-06-29 Attenuation efficace de pre-echos dans un signal audionumerique
PCT/FR2013/051517 WO2014001730A1 (fr) 2012-06-29 2013-06-28 Atténuation efficace de pré-échos dans un signal audionumérique

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BR112014032587B1 (pt) 2022-08-09
BR112014032587A2 (pt) 2017-06-27
KR102082156B1 (ko) 2020-04-14
WO2014001730A1 (fr) 2014-01-03
MX2014015065A (es) 2015-02-17
CN104395958B (zh) 2017-09-05
EP2867893A1 (de) 2015-05-06
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CN104395958A (zh) 2015-03-04
US9489964B2 (en) 2016-11-08
RU2607418C2 (ru) 2017-01-10
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