This application is a U.S. National Phase Application under 35 U.S.C. 371 of International Application No. PCT/FR2011/051425 filed Jun. 21, 2011, which was published on Dec. 29, 2011 under International Publication Number WO 2011/161372 A1, which claims the benefit of French Patent Application No. 1002674 filed on Jun. 25, 2010. The disclosures of these applications are incorporated herein by reference in their entirety.
The invention relates to real-time audio digital synthesis.
More particularly, the invention is applicable to fields that require performing a spectral modification of an audio signal in real-time, for example in hearing aids, video games, public address systems, etc.
Since the mid-1980s, solutions have been sought to make it possible to reconstruct an audio signal from its amplitude spectrum. To that end, these methods are based on the use of the short-term Fourier transform (STFT) under specific conditions.
These methods nevertheless have several drawbacks. First, they require working on the entire signal, which prevents working in real-time. Furthermore, these methods are based on an unbounded iterative reduction, which has a high computation cost.
The invention aims to improve this situation.
To that end, the invention proposes an audio digital synthesizer, comprising:
In this synthesizer:
Other features and advantages of the invention will better appear upon reading the following description, taken from examples provided as an illustration and non-limitingly, taken from the drawings, in which:
The detailed description is supplemented by annex A, which provides certain mathematical formulas that are used in the context of the invention. This annex is set apart for clarity reasons, and to facilitate references thereto. It is an integral part of the description, and may therefore be used not only to better understand the present invention, but also to contribute to the definition thereof, if applicable.
The drawings and the description below essentially contain elements of a definite nature. They may therefore not only be used to better understand the present invention, but also to contribute to the definition thereof, if applicable.
During operation, the synthesizer 100 receives, as input, digital data 150 (hereafter denoted Fi(ω)) representing the amplitude spectrum of an audio signal. The digital data 150 is processed by the synthesizer 100, and is returned in the form of digital audio signal data 160.
As shown in
Each frame comprises N samples, the last L of which are extrapolated. L is selected in the example described here as being a divisor of the size N of the STFT. Preferably, N is a multiple of 4 L. The audio synthesizer 100 operates based on a loop, in which a frame is extrapolated upon each iteration.
The synthesizer 100 operates according to the flowchart of
In a first operation 200, the audio synthesis is initialized using a function Init( ). The function Init( ) prepares the variables and parameters that will be used in the synthesis and assigns them original values. In the example described here, the function Init( ) is performed by the dialer 130.
These variables and parameters are:
Once all of the variables have been initialized, the resynthesis loop begins at 210 with the incrementation of the counter i. Then, a test on the value of the counter i in an operation 220 may trigger a loop exit operation 230 if all of the frames have been processed.
Otherwise, draft data y(n) is calculated in an operation 240 using a function Ext( ). In the example described here, the function Ext( ) is performed by the extrapolator 110. The function Ext( ) uses the auxiliary data xi−1(n) as argument.
For all of the iterations, the auxiliary data x1−1(n) is calculated in the previous iteration, except for the first, where x0(n) is calculated by the function Init( ).
For each n varying between 0 and N−L−1 , the function Ext( ) defines y(n) as the product of the data xi−1(n+L) by the square of a window function H(n) according to formula (10) provided in annex A. For each n varying between N−L and N−1, the function Ext( ) defines y(n) equal to 0.
Then, the computer 120 is called upon in an operation 250 to calculate the discrete Fourier transform of y(n) using a function DFT( ). The result of this function is stored in data Y(w) where w varies between 0 and N−1 , according to formula (20) of annex A.
The dialer 130 is then called upon in an operation 260 to combine the data Y(w) with data Fi(w) to produce data Z(w), using a function Comp( ).
The data Fi(w) is amplitude spectrum data of the signal to be resynthesized corresponding to the frame of the current iteration. In the example described here, the data Fi(w) is calculated by STFT of the signal to be resynthesized, using the function H(n) as the window function.
The function Comp( ) combines the data Y(w) with the data Fi(w) by applying the module Fi(w) to the data Y(w). Thus, in the data Z(w), the draft data phase y(n) is kept, and the energy from the frame to be resynthesized is reintroduced.
In the example described here, the data Fi(w) is received as input. However, the synthesizer 100 may comprise an audio spectrometer to calculate these coefficients and provide them to the dialer 130.
The synthesizer may also comprise a component placed between the spectrometer and the input memory capable of modifying the digital data representative of the input amplitude spectrum, and to transmit the modified amplitude spectrum data to the input memory.
This component can apply any preprocessing algorithm or filter to the amplitude spectrum, for example in the context of a transformation of the type transforming the esophageal voice into the laryngeal voice.
The data Z(w) is then returned to the computer 120 in an operation 270. In that operation, the computer 120 executes a function IDFT( ) that performs the inverse of the operation 250, and stores the result in data z(n), according to formula (30) of annex A.
The dialer 130 is then called upon again in an operation 280 to calculate the auxiliary data xi(n) for the following iteration. This operation is done using a function Rest( ) that receives the data z(n) as input, and divides it by the window function H(n).
Lastly, the resynthesized signal s(n) is lastly calculated in an operation 290 using a function Add( ), then the loop resumes in 210. The function Add( ) is carried out by the adder 140, and receives the data z(n) and the counter i as arguments. The function Add( ) then adds the data z(n) multiplied by the window function H(n) to the resynthesized data with index s(i*L+n), according to formula (40) of annex A.
As was mentioned in the introduction, the field of resynthesis from the amplitude spectrum is a field that is poorly known and difficult to understand. This field calls for mastery of short-term Fourier transforms (STFT), the physical meaning of which has still not been mastered.
Although the phenomena in question are not fully understood, it is possible to provide at least a partial explanation of them, as will be seen now.
Each iteration makes it possible to extrapolate the L samples of a current frame. Since these samples are the continuation of the preceding frames, the auxiliary data xi−1(n) therefore already contains a large quantity of the signal.
That is why the auxiliary data x−1(n) is first shifted to the left by L indices, and the last L elements are left zero. Then, the draft data used to calculate the resynthesis data is multiplied by the window function H(n) squared. The window function used in the example described here is a normalized Hamming window, the formula for which is provided in (50) in annex A.
In fact, the operation 250 is a modified STFT, because it is the square of the window function that is applied instead of the window function. That is why the data z(n) is divided by the window function H(n) to give the auxiliary data xi(n) in the operation 280.
The multiplications and divisions by the window function are not done randomly. In fact, it is possible to consider eliminating the division from the operation 280 and performing a simple multiplication in the operation 240 instead of multiplication by the square.
But this would not account for the fact that the multiplication of the operation 250 is done over the x−1(n) offset relative to those of the operation 280. And this detail is crucial, as it makes it possible to use a signal xi(n) that is a sort of “idealized” vision of the resynthesized audio signal.
Conversely, the resynthesized data s(n) is windowed relative to the data z(n). This is done so as to obtain a smoothing effect of the overlap-add (OLA) type, and makes it possible to limit discontinuities at the ends of the frames.
The annex A and these explanations are not just a series of theoretical mathematical formulas. Thus, the Applicant first used a formula (50) with a simple multiplication instead of multiplication by the square.
The experiments not being satisfactory, its research led it to use the square of that window. This is advantageous because multiplication by the square of the window function ensures normalization and saves one computation step.
Other functions could be used for the window function, such as the normalized Hann window or another normalized window. The use of the normalized window is important because it makes it possible on the one hand to smooth the resynthesized data s(n) without it being necessary on the other hand to normalize the latter at the end of resynthesis.
In fact, without smoothing, artifacts would appear at the borders of the frames. And without a normalized window, it would be necessary to take all of the produced elements into account to normalize, which would prevent a real-time application.
Number | Date | Country | Kind |
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10 02674 | Jun 2010 | FR | national |
Filing Document | Filing Date | Country | Kind | 371c Date |
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PCT/FR2011/051425 | 6/21/2011 | WO | 00 | 12/24/2012 |
Publishing Document | Publishing Date | Country | Kind |
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WO2011/161372 | 12/29/2011 | WO | A |
Number | Name | Date | Kind |
---|---|---|---|
4922797 | Chapman | May 1990 | A |
5327521 | Savic et al. | Jul 1994 | A |
5401897 | Depalle et al. | Mar 1995 | A |
5504833 | George et al. | Apr 1996 | A |
5998725 | Ohta | Dec 1999 | A |
6526376 | Villette et al. | Feb 2003 | B1 |
6992245 | Kenmochi et al. | Jan 2006 | B2 |
7257230 | Nagatani | Aug 2007 | B2 |
7685218 | McGrath | Mar 2010 | B2 |
20020007273 | Chen | Jan 2002 | A1 |
20030009336 | Kenmochi et al. | Jan 2003 | A1 |
20030187635 | Ramabadran et al. | Oct 2003 | A1 |
20040133423 | Crockett | Jul 2004 | A1 |
20050010397 | Sakurai et al. | Jan 2005 | A1 |
20050065784 | McAulay et al. | Mar 2005 | A1 |
20070282203 | Baba et al. | Dec 2007 | A1 |
20080046252 | Zopf et al. | Feb 2008 | A1 |
20110087488 | Morinaka et al. | Apr 2011 | A1 |
20110213476 | Eisenberg | Sep 2011 | A1 |
20110288873 | Nagel et al. | Nov 2011 | A1 |
20110320207 | Rodriguez Crespo et al. | Dec 2011 | A1 |
20120008799 | Disch et al. | Jan 2012 | A1 |
20120010882 | Thyssen et al. | Jan 2012 | A1 |
20120150544 | McLoughlin et al. | Jun 2012 | A1 |
20130044896 | Ekstrand | Feb 2013 | A1 |
Entry |
---|
Maher, Robert C. “A Method for Extrapolation of Missing Digital Audio Data”. Oct. 1993. AES. |
International Search Report of the International Searching Authority mailed on Oct. 13, 2011, issued in connection with International Patent Appln. No. PCT/FR2011/051425 (5 pages). |
Griffin, et al., “Signal Estimation from Modified Short-Time Fourier Transform,” IEEE Transactions on Acoustics, Speech, and Signal Processing, vol. ASSP-32, No. 2, Apr. 1984 (8 pages). |
Crochiere, “A Weighted Overlap-Add Method of Short-Tine Fourier Analysis/Synthesis,” IEEE Transactions on Acoustics, Speech, and Signal Processing, vol. ASSP-28, No. 1, Feb. 1980 (4 pages). |
Number | Date | Country | |
---|---|---|---|
20130103173 A1 | Apr 2013 | US |